diff options
Diffstat (limited to 'sound')
55 files changed, 2036 insertions, 601 deletions
diff --git a/sound/oss/ac97_codec.c b/sound/oss/ac97_codec.c index b63839e8f9bd..456a1b4d7832 100644 --- a/sound/oss/ac97_codec.c +++ b/sound/oss/ac97_codec.c @@ -30,7 +30,7 @@ ************************************************************************** * * History - * May 02, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * May 02, 2003 Liam Girdwood <lrg@slimlogic.co.uk> * Removed non existant WM9700 * Added support for WM9705, WM9708, WM9709, WM9710, WM9711 * WM9712 and WM9717 diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 6ce3cbe98a6a..6e831aff1bd0 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -476,7 +476,7 @@ static int patch_yamaha_ymf753(struct snd_ac97 * ac97) } /* - * May 2, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * May 2, 2003 Liam Girdwood <lrg@slimlogic.co.uk> * removed broken wolfson00 patch. * added support for WM9705,WM9708,WM9709,WM9710,WM9711,WM9712 and WM9717. */ diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index c461baa83c2a..c59065513118 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -322,8 +322,8 @@ static hda_nid_t stac92hd71bxx_mux_nids[2] = { 0x1a, 0x1b }; -static hda_nid_t stac92hd71bxx_dmux_nids[1] = { - 0x1c, +static hda_nid_t stac92hd71bxx_dmux_nids[2] = { + 0x1c, 0x1d, }; static hda_nid_t stac92hd71bxx_smux_nids[2] = { @@ -861,20 +861,18 @@ static struct hda_verb stac92hd71bxx_core_init[] = { { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, }; -#define HD_DISABLE_PORTF 3 +#define HD_DISABLE_PORTF 2 static struct hda_verb stac92hd71bxx_analog_core_init[] = { /* start of config #1 */ /* connect port 0f to audio mixer */ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ /* unmute right and left channels for node 0x0f */ { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* start of config #2 */ @@ -883,10 +881,6 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = { { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* connect port 0d to audio mixer */ - { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, - /* unmute dac0 input in audio mixer */ - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, /* unmute right and left channels for nodes 0x0a, 0xd */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -1107,6 +1101,7 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = { static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { STAC_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1119,8 +1114,17 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT), */ - HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x0, HDA_INPUT), + + HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x1, HDA_INPUT), + + HDA_CODEC_MUTE("DAC0 Capture Switch", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x17, 0x3, HDA_INPUT), + + HDA_CODEC_MUTE("DAC1 Capture Switch", 0x17, 0x4, HDA_INPUT), + HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x17, 0x4, HDA_INPUT), { } /* end */ }; @@ -1649,7 +1653,7 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { static unsigned int ref92hd71bxx_pin_configs[11] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, - 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, + 0x0181302e, 0x01014010, 0x01019020, 0x90a000f0, 0x90a000f0, 0x01452050, 0x01452050, }; @@ -3000,7 +3004,7 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec) /* labels for amp mux outputs */ static const char *stac92xx_amp_labels[3] = { - "Front Microphone", "Microphone", "Line In" + "Front Microphone", "Microphone", "Line In", }; /* create amp out controls mux on capable codecs */ @@ -4327,6 +4331,16 @@ static struct hda_codec_ops stac92hd71bxx_patch_ops = { #endif }; +static struct hda_input_mux stac92hd71bxx_dmux = { + .num_items = 4, + .items = { + { "Analog Inputs", 0x00 }, + { "Mixer", 0x01 }, + { "Digital Mic 1", 0x02 }, + { "Digital Mic 2", 0x03 }, + } +}; + static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -4341,6 +4355,8 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pin_nids = stac92hd71bxx_pin_nids; + memcpy(&spec->private_dimux, &stac92hd71bxx_dmux, + sizeof(stac92hd71bxx_dmux)); spec->board_config = snd_hda_check_board_config(codec, STAC_92HD71BXX_MODELS, stac92hd71bxx_models, @@ -4392,6 +4408,7 @@ again: /* no output amps */ spec->num_pwrs = 0; spec->mixer = stac92hd71bxx_analog_mixer; + spec->dinput_mux = &spec->private_dimux; /* disable VSW */ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; @@ -4409,12 +4426,13 @@ again: spec->num_pwrs = 0; /* fallthru */ default: + spec->dinput_mux = &spec->private_dimux; spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; } - spec->aloopback_mask = 0x20; + spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; if (spec->board_config > STAC_92HD71BXX_REF) { @@ -4456,6 +4474,10 @@ again: spec->multiout.num_dacs = 1; spec->multiout.hp_nid = 0x11; spec->multiout.dac_nids = stac92hd71bxx_dac_nids; + if (spec->dinput_mux) + spec->private_dimux.num_items += + spec->num_dmics - + (ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1); err = stac92xx_parse_auto_config(codec, 0x21, 0x23); if (!err) { diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig index 905186502e00..85a883299c2e 100644 --- a/sound/soc/at91/Kconfig +++ b/sound/soc/at91/Kconfig @@ -8,20 +8,3 @@ config SND_AT91_SOC config SND_AT91_SOC_SSC tristate - -config SND_AT91_SOC_ETI_B1_WM8731 - tristate "SoC Audio support for WM8731-based Endrelia ETI-B1 boards" - depends on SND_AT91_SOC && (MACH_ETI_B1 || MACH_ETI_C1) - select SND_AT91_SOC_SSC - select SND_SOC_WM8731 - help - Say Y if you want to add support for SoC audio on WM8731-based - Endrelia Technologies Inc ETI-B1 or ETI-C1 boards. - -config SND_AT91_SOC_ETI_SLAVE - bool "Run codec in slave Mode on Endrelia boards" - depends on SND_AT91_SOC_ETI_B1_WM8731 - default n - help - Say Y if you want to run with the AT91 SSC generating the BCLK - and LRC signals on Endrelia boards. diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile index f23da17cc328..b817f11df286 100644 --- a/sound/soc/at91/Makefile +++ b/sound/soc/at91/Makefile @@ -4,8 +4,3 @@ snd-soc-at91-ssc-objs := at91-ssc.o obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o - -# AT91 Machine Support -snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o - -obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c index a5b1a79ebffb..1b61cc461261 100644 --- a/sound/soc/at91/at91-ssc.c +++ b/sound/soc/at91/at91-ssc.c @@ -5,7 +5,7 @@ * Endrelia Technologies Inc. * * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Liam Girdwood <lrg@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 diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c deleted file mode 100644 index 684781e4088b..000000000000 --- a/sound/soc/at91/eti_b1_wm8731.c +++ /dev/null @@ -1,349 +0,0 @@ -/* - * eti_b1_wm8731 -- SoC audio for AT91RM9200-based Endrelia ETI_B1 board. - * - * Author: Frank Mandarino <fmandarino@endrelia.com> - * Endrelia Technologies Inc. - * Created: Mar 29, 2006 - * - * Based on corgi.c by: - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * - * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> - * Richard Purdie <richard@openedhand.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/kernel.h> -#include <linux/clk.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <mach/hardware.h> -#include <mach/gpio.h> - -#include "../codecs/wm8731.h" -#include "at91-pcm.h" -#include "at91-ssc.h" - -#if 0 -#define DBG(x...) printk(KERN_INFO "eti_b1_wm8731: " x) -#else -#define DBG(x...) -#endif - -static struct clk *pck1_clk; -static struct clk *pllb_clk; - - -static int eti_b1_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - int ret; - - /* cpu clock is the AT91 master clock sent to the SSC */ - ret = snd_soc_dai_set_sysclk(cpu_dai, AT91_SYSCLK_MCK, - 60000000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* codec system clock is supplied by PCK1, set to 12MHz */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, - 12000000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* Start PCK1 clock. */ - clk_enable(pck1_clk); - DBG("pck1 started\n"); - - return 0; -} - -static void eti_b1_shutdown(struct snd_pcm_substream *substream) -{ - /* Stop PCK1 clock. */ - clk_disable(pck1_clk); - DBG("pck1 stopped\n"); -} - -static int eti_b1_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->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - int ret; - -#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE - unsigned int rate; - int cmr_div, period; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - 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_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* - * The SSC clock dividers depend on the sample rate. The CMR.DIV - * field divides the system master clock MCK to drive the SSC TK - * signal which provides the codec BCLK. The TCMR.PERIOD and - * RCMR.PERIOD fields further divide the BCLK signal to drive - * the SSC TF and RF signals which provide the codec DACLRC and - * ADCLRC clocks. - * - * The dividers were determined through trial and error, where a - * CMR.DIV value is chosen such that the resulting BCLK value is - * divisible, or almost divisible, by (2 * sample rate), and then - * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. - */ - rate = params_rate(params); - - switch (rate) { - case 8000: - cmr_div = 25; /* BCLK = 60MHz/(2*25) = 1.2MHz */ - period = 74; /* LRC = BCLK/(2*(74+1)) = 8000Hz */ - break; - case 32000: - cmr_div = 7; /* BCLK = 60MHz/(2*7) ~= 4.28571428MHz */ - period = 66; /* LRC = BCLK/(2*(66+1)) = 31982.942Hz */ - break; - case 48000: - cmr_div = 13; /* BCLK = 60MHz/(2*13) ~= 2.3076923MHz */ - period = 23; /* LRC = BCLK/(2*(23+1)) = 48076.923Hz */ - break; - default: - printk(KERN_WARNING "unsupported rate %d on ETI-B1 board\n", rate); - return -EINVAL; - } - - /* set the MCK divider for BCLK */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); - if (ret < 0) - return ret; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* set the BCLK divider for DACLRC */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - AT91SSC_TCMR_PERIOD, period); - } else { - /* set the BCLK divider for ADCLRC */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - AT91SSC_RCMR_PERIOD, period); - } - if (ret < 0) - return ret; - -#else /* CONFIG_SND_AT91_SOC_ETI_SLAVE */ - /* - * Codec in Master Mode. - */ - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - -#endif /* CONFIG_SND_AT91_SOC_ETI_SLAVE */ - - return 0; -} - -static struct snd_soc_ops eti_b1_ops = { - .startup = eti_b1_startup, - .hw_params = eti_b1_hw_params, - .shutdown = eti_b1_shutdown, -}; - - -static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), -}; - -static const struct snd_soc_dapm_route intercon[] = { - - /* speaker connected to LHPOUT */ - {"Ext Spk", NULL, "LHPOUT"}, - - /* mic is connected to Mic Jack, with WM8731 Mic Bias */ - {"MICIN", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Int Mic"}, -}; - -/* - * Logic for a wm8731 as connected on a Endrelia ETI-B1 board. - */ -static int eti_b1_wm8731_init(struct snd_soc_codec *codec) -{ - DBG("eti_b1_wm8731_init() called\n"); - - /* Add specific widgets */ - snd_soc_dapm_new_controls(codec, eti_b1_dapm_widgets, - ARRAY_SIZE(eti_b1_dapm_widgets)); - - /* Set up specific audio path interconnects */ - snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon)); - - /* not connected */ - snd_soc_dapm_disable_pin(codec, "RLINEIN"); - snd_soc_dapm_disable_pin(codec, "LLINEIN"); - - /* always connected */ - snd_soc_dapm_enable_pin(codec, "Int Mic"); - snd_soc_dapm_enable_pin(codec, "Ext Spk"); - - snd_soc_dapm_sync(codec); - - return 0; -} - -static struct snd_soc_dai_link eti_b1_dai = { - .name = "WM8731", - .stream_name = "WM8731 PCM", - .cpu_dai = &at91_ssc_dai[1], - .codec_dai = &wm8731_dai, - .init = eti_b1_wm8731_init, - .ops = &eti_b1_ops, -}; - -static struct snd_soc_machine snd_soc_machine_eti_b1 = { - .name = "ETI_B1_WM8731", - .dai_link = &eti_b1_dai, - .num_links = 1, -}; - -static struct wm8731_setup_data eti_b1_wm8731_setup = { - .i2c_bus = 0, - .i2c_address = 0x1a, -}; - -static struct snd_soc_device eti_b1_snd_devdata = { - .machine = &snd_soc_machine_eti_b1, - .platform = &at91_soc_platform, - .codec_dev = &soc_codec_dev_wm8731, - .codec_data = &eti_b1_wm8731_setup, -}; - -static struct platform_device *eti_b1_snd_device; - -static int __init eti_b1_init(void) -{ - int ret; - struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data; - - if (!request_mem_region(AT91RM9200_BASE_SSC1, SZ_16K, "soc-audio")) { - DBG("SSC1 memory region is busy\n"); - return -EBUSY; - } - - ssc->base = ioremap(AT91RM9200_BASE_SSC1, SZ_16K); - if (!ssc->base) { - DBG("SSC1 memory ioremap failed\n"); - ret = -ENOMEM; - goto fail_release_mem; - } - - ssc->pid = AT91RM9200_ID_SSC1; - - eti_b1_snd_device = platform_device_alloc("soc-audio", -1); - if (!eti_b1_snd_device) { - DBG("platform device allocation failed\n"); - ret = -ENOMEM; - goto fail_io_unmap; - } - - platform_set_drvdata(eti_b1_snd_device, &eti_b1_snd_devdata); - eti_b1_snd_devdata.dev = &eti_b1_snd_device->dev; - - ret = platform_device_add(eti_b1_snd_device); - if (ret) { - DBG("platform device add failed\n"); - platform_device_put(eti_b1_snd_device); - goto fail_io_unmap; - } - - at91_set_A_periph(AT91_PIN_PB6, 0); /* TF1 */ - at91_set_A_periph(AT91_PIN_PB7, 0); /* TK1 */ - at91_set_A_periph(AT91_PIN_PB8, 0); /* TD1 */ - at91_set_A_periph(AT91_PIN_PB9, 0); /* RD1 */ -/* at91_set_A_periph(AT91_PIN_PB10, 0);*/ /* RK1 */ - at91_set_A_periph(AT91_PIN_PB11, 0); /* RF1 */ - - /* - * Set PCK1 parent to PLLB and its rate to 12 Mhz. - */ - pllb_clk = clk_get(NULL, "pllb"); - pck1_clk = clk_get(NULL, "pck1"); - - clk_set_parent(pck1_clk, pllb_clk); - clk_set_rate(pck1_clk, 12000000); - - DBG("MCLK rate %luHz\n", clk_get_rate(pck1_clk)); - - /* assign the GPIO pin to PCK1 */ - at91_set_B_periph(AT91_PIN_PA24, 0); - -#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE - printk(KERN_INFO "eti_b1_wm8731: Codec in Slave Mode\n"); -#else - printk(KERN_INFO "eti_b1_wm8731: Codec in Master Mode\n"); -#endif - return ret; - -fail_io_unmap: - iounmap(ssc->base); -fail_release_mem: - release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K); - return ret; -} - -static void __exit eti_b1_exit(void) -{ - struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data; - - clk_put(pck1_clk); - clk_put(pllb_clk); - - platform_device_unregister(eti_b1_snd_device); - - iounmap(ssc->base); - release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K); -} - -module_init(eti_b1_init); -module_exit(eti_b1_exit); - -/* Module information */ -MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>"); -MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index f98331d099e7..dc006206f622 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -17,6 +17,22 @@ config SND_BF5XX_SOC_SSM2602 help Say Y if you want to add support for SoC audio on BF527-EZKIT. +config SND_BF5XX_SOC_AD73311 + tristate "SoC AD73311 Audio support for Blackfin" + depends on SND_BF5XX_I2S + select SND_BF5XX_SOC_I2S + select SND_SOC_AD73311 + help + Say Y if you want to add support for AD73311 codec on Blackfin. + +config SND_BFIN_AD73311_SE + int "PF pin for AD73311L Chip Select" + depends on SND_BF5XX_SOC_AD73311 + default 4 + help + Enter the GPIO used to control AD73311's SE pin. Acceptable + values are 0 to 7 + config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" depends on BLACKFIN && SND_SOC diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 9ea8bd9e0ba3..97bb37a6359c 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -14,7 +14,8 @@ obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o # Blackfin Machine Support snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o - +snd-ad73311-objs := bf5xx-ad73311.o obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o +obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 51f4907c4831..25e50d2ea1ec 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -56,6 +56,7 @@ static void bf5xx_mmap_copy(struct snd_pcm_substream *substream, sport->tx_pos += runtime->period_size; if (sport->tx_pos >= runtime->buffer_size) sport->tx_pos %= runtime->buffer_size; + sport->tx_delay_pos = sport->tx_pos; } else { bf5xx_ac97_to_pcm( (struct ac97_frame *)sport->rx_dma_buf + sport->rx_pos, @@ -72,7 +73,15 @@ static void bf5xx_dma_irq(void *data) struct snd_pcm_substream *pcm = data; #if defined(CONFIG_SND_MMAP_SUPPORT) struct snd_pcm_runtime *runtime = pcm->runtime; + struct sport_device *sport = runtime->private_data; bf5xx_mmap_copy(pcm, runtime->period_size); + if (pcm->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (sport->once == 0) { + snd_pcm_period_elapsed(pcm); + bf5xx_mmap_copy(pcm, runtime->period_size); + sport->once = 1; + } + } #endif snd_pcm_period_elapsed(pcm); } @@ -114,6 +123,10 @@ static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + memset(runtime->dma_area, 0, runtime->buffer_size); snd_pcm_lib_free_pages(substream); return 0; } @@ -127,16 +140,11 @@ static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) * SPORT working in TMD mode(include AC97). */ #if defined(CONFIG_SND_MMAP_SUPPORT) - size_t size = bf5xx_pcm_hardware.buffer_bytes_max - * sizeof(struct ac97_frame) / 4; - /*clean up intermediate buffer*/ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - memset(sport->tx_dma_buf, 0, size); sport_set_tx_callback(sport, bf5xx_dma_irq, substream); sport_config_tx_dma(sport, sport->tx_dma_buf, runtime->periods, runtime->period_size * sizeof(struct ac97_frame)); } else { - memset(sport->rx_dma_buf, 0, size); sport_set_rx_callback(sport, bf5xx_dma_irq, substream); sport_config_rx_dma(sport, sport->rx_dma_buf, runtime->periods, runtime->period_size * sizeof(struct ac97_frame)); @@ -164,8 +172,12 @@ static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) pr_debug("%s enter\n", __func__); switch (cmd) { case SNDRV_PCM_TRIGGER_START: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + bf5xx_mmap_copy(substream, runtime->period_size); + snd_pcm_period_elapsed(substream); + sport->tx_delay_pos = 0; sport_tx_start(sport); + } else sport_rx_start(sport); break; @@ -198,7 +210,7 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) #if defined(CONFIG_SND_MMAP_SUPPORT) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - curr = sport->tx_pos; + curr = sport->tx_delay_pos; else curr = sport->rx_pos; #else @@ -237,6 +249,21 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream) return ret; } +static int bf5xx_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + + pr_debug("%s enter\n", __func__); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + sport->once = 0; + memset(sport->tx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame)); + } else + memset(sport->rx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame)); + + return 0; +} + #ifdef CONFIG_SND_MMAP_SUPPORT static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) @@ -272,6 +299,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, struct snd_pcm_ops bf5xx_pcm_ac97_ops = { .open = bf5xx_pcm_open, + .close = bf5xx_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = bf5xx_pcm_hw_params, .hw_free = bf5xx_pcm_hw_free, diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index c782e311fd56..5e5aafb6485f 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -129,7 +129,6 @@ static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data) struct ac97_frame *nextwrite; sport_incfrag(sport, &nextfrag, 1); - sport_incfrag(sport, &nextfrag, 1); nextwrite = (struct ac97_frame *)(sport->tx_buf + \ nextfrag * sport->tx_fragsize); diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c new file mode 100644 index 000000000000..622c9b909532 --- /dev/null +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -0,0 +1,240 @@ +/* + * File: sound/soc/blackfin/bf5xx-ad73311.c + * Author: Cliff Cai <Cliff.Cai@analog.com> + * + * Created: Thur Sep 25 2008 + * Description: Board driver for ad73311 sound chip + * + * Modified: + * Copyright 2008 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> + +#include <asm/blackfin.h> +#include <asm/cacheflush.h> +#include <asm/irq.h> +#include <asm/dma.h> +#include <asm/portmux.h> + +#include "../codecs/ad73311.h" +#include "bf5xx-sport.h" +#include "bf5xx-i2s-pcm.h" +#include "bf5xx-i2s.h" + +#if CONFIG_SND_BF5XX_SPORT_NUM == 0 +#define bfin_write_SPORT_TCR1 bfin_write_SPORT0_TCR1 +#define bfin_read_SPORT_TCR1 bfin_read_SPORT0_TCR1 +#define bfin_write_SPORT_TCR2 bfin_write_SPORT0_TCR2 +#define bfin_write_SPORT_TX16 bfin_write_SPORT0_TX16 +#define bfin_read_SPORT_STAT bfin_read_SPORT0_STAT +#else +#define bfin_write_SPORT_TCR1 bfin_write_SPORT1_TCR1 +#define bfin_read_SPORT_TCR1 bfin_read_SPORT1_TCR1 +#define bfin_write_SPORT_TCR2 bfin_write_SPORT1_TCR2 +#define bfin_write_SPORT_TX16 bfin_write_SPORT1_TX16 +#define bfin_read_SPORT_STAT bfin_read_SPORT1_STAT +#endif + +#define GPIO_SE CONFIG_SND_BFIN_AD73311_SE + +static struct snd_soc_machine bf5xx_ad73311; + +static int snd_ad73311_startup(void) +{ + pr_debug("%s enter\n", __func__); + + /* Pull up SE pin on AD73311L */ + gpio_set_value(GPIO_SE, 1); + return 0; +} + +static int snd_ad73311_configure(void) +{ + unsigned short ctrl_regs[6]; + unsigned short status = 0; + int count = 0; + + /* DMCLK = MCLK = 16.384 MHz + * SCLK = DMCLK/8 = 2.048 MHz + * Sample Rate = DMCLK/2048 = 8 KHz + */ + ctrl_regs[0] = AD_CONTROL | AD_WRITE | CTRL_REG_B | REGB_MCDIV(0) | \ + REGB_SCDIV(0) | REGB_DIRATE(0); + ctrl_regs[1] = AD_CONTROL | AD_WRITE | CTRL_REG_C | REGC_PUDEV | \ + REGC_PUADC | REGC_PUDAC | REGC_PUREF | REGC_REFUSE ; + ctrl_regs[2] = AD_CONTROL | AD_WRITE | CTRL_REG_D | REGD_OGS(2) | \ + REGD_IGS(2); + ctrl_regs[3] = AD_CONTROL | AD_WRITE | CTRL_REG_E | REGE_DA(0x1f); + ctrl_regs[4] = AD_CONTROL | AD_WRITE | CTRL_REG_F | REGF_SEEN ; + ctrl_regs[5] = AD_CONTROL | AD_WRITE | CTRL_REG_A | REGA_MODE_DATA; + + local_irq_disable(); + snd_ad73311_startup(); + udelay(1); + + bfin_write_SPORT_TCR1(TFSR); + bfin_write_SPORT_TCR2(0xF); + SSYNC(); + + /* SPORT Tx Register is a 8 x 16 FIFO, all the data can be put to + * FIFO before enable SPORT to transfer the data + */ + for (count = 0; count < 6; count++) + bfin_write_SPORT_TX16(ctrl_regs[count]); + SSYNC(); + bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() | TSPEN); + SSYNC(); + + /* When TUVF is set, the data is already send out */ + while (!(status & TUVF) && count++ < 10000) { + udelay(1); + status = bfin_read_SPORT_STAT(); + SSYNC(); + } + bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() & ~TSPEN); + SSYNC(); + local_irq_enable(); + + if (count == 10000) { + printk(KERN_ERR "ad73311: failed to configure codec\n"); + return -1; + } + return 0; +} + +static int bf5xx_probe(struct platform_device *pdev) +{ + int err; + if (gpio_request(GPIO_SE, "AD73311_SE")) { + printk(KERN_ERR "%s: Failed ro request GPIO_%d\n", __func__, GPIO_SE); + return -EBUSY; + } + + gpio_direction_output(GPIO_SE, 0); + + err = snd_ad73311_configure(); + if (err < 0) + return -EFAULT; + + return 0; +} + +static int bf5xx_ad73311_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + pr_debug("%s enter\n", __func__); + cpu_dai->private_data = sport_handle; + return 0; +} + +static int bf5xx_ad73311_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 *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + + pr_debug("%s rate %d format %x\n", __func__, params_rate(params), + params_format(params)); + + /* set cpu DAI configuration */ + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + return 0; +} + + +static struct snd_soc_ops bf5xx_ad73311_ops = { + .startup = bf5xx_ad73311_startup, + .hw_params = bf5xx_ad73311_hw_params, +}; + +static struct snd_soc_dai_link bf5xx_ad73311_dai = { + .name = "ad73311", + .stream_name = "AD73311", + .cpu_dai = &bf5xx_i2s_dai, + .codec_dai = &ad73311_dai, + .ops = &bf5xx_ad73311_ops, +}; + +static struct snd_soc_machine bf5xx_ad73311 = { + .name = "bf5xx_ad73311", + .probe = bf5xx_probe, + .dai_link = &bf5xx_ad73311_dai, + .num_links = 1, +}; + +static struct snd_soc_device bf5xx_ad73311_snd_devdata = { + .machine = &bf5xx_ad73311, + .platform = &bf5xx_i2s_soc_platform, + .codec_dev = &soc_codec_dev_ad73311, +}; + +static struct platform_device *bf52x_ad73311_snd_device; + +static int __init bf5xx_ad73311_init(void) +{ + int ret; + + pr_debug("%s enter\n", __func__); + bf52x_ad73311_snd_device = platform_device_alloc("soc-audio", -1); + if (!bf52x_ad73311_snd_device) + return -ENOMEM; + + platform_set_drvdata(bf52x_ad73311_snd_device, &bf5xx_ad73311_snd_devdata); + bf5xx_ad73311_snd_devdata.dev = &bf52x_ad73311_snd_device->dev; + ret = platform_device_add(bf52x_ad73311_snd_device); + + if (ret) + platform_device_put(bf52x_ad73311_snd_device); + + return ret; +} + +static void __exit bf5xx_ad73311_exit(void) +{ + pr_debug("%s enter\n", __func__); + platform_device_unregister(bf52x_ad73311_snd_device); +} + +module_init(bf5xx_ad73311_init); +module_exit(bf5xx_ad73311_exit); + +/* Module information */ +MODULE_AUTHOR("Cliff Cai"); +MODULE_DESCRIPTION("ALSA SoC AD73311 Blackfin"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 43a4092eeb89..827587f08180 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -70,6 +70,13 @@ static struct sport_param sport_params[2] = { } }; +static u16 sport_req[][7] = { + { P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, + P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}, + { P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, + P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}, +}; + static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -78,6 +85,14 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* interface format:support I2S,slave mode */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + bf5xx_i2s.tcr1 |= TFSR | TCKFE; + bf5xx_i2s.rcr1 |= RFSR | RCKFE; + bf5xx_i2s.tcr2 |= TSFSE; + bf5xx_i2s.rcr2 |= RSFSE; + break; + case SND_SOC_DAIFMT_DSP_A: + bf5xx_i2s.tcr1 |= TFSR; + bf5xx_i2s.rcr1 |= RFSR; break; case SND_SOC_DAIFMT_LEFT_J: ret = -EINVAL; @@ -127,14 +142,17 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: bf5xx_i2s.tcr2 |= 15; bf5xx_i2s.rcr2 |= 15; + sport_handle->wdsize = 2; break; case SNDRV_PCM_FORMAT_S24_LE: bf5xx_i2s.tcr2 |= 23; bf5xx_i2s.rcr2 |= 23; + sport_handle->wdsize = 3; break; case SNDRV_PCM_FORMAT_S32_LE: bf5xx_i2s.tcr2 |= 31; bf5xx_i2s.rcr2 |= 31; + sport_handle->wdsize = 4; break; } @@ -145,17 +163,17 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, * need to configure both of them at the time when the first * stream is opened. * - * CPU DAI format:I2S, slave mode. + * CPU DAI:slave mode. */ - ret = sport_config_rx(sport_handle, RFSR | RCKFE, - RSFSE|bf5xx_i2s.rcr2, 0, 0); + ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1, + bf5xx_i2s.rcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - ret = sport_config_tx(sport_handle, TFSR | TCKFE, - TSFSE|bf5xx_i2s.tcr2, 0, 0); + ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1, + bf5xx_i2s.tcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; @@ -174,13 +192,6 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream) static int bf5xx_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { - u16 sport_req[][7] = { - { P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, - P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}, - { P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, - P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}, - }; - pr_debug("%s enter\n", __func__); if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) { pr_err("Requesting Peripherals failed\n"); @@ -198,6 +209,13 @@ static int bf5xx_i2s_probe(struct platform_device *pdev, return 0; } +static void bf5xx_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + pr_debug("%s enter\n", __func__); + peripheral_free_list(&sport_req[sport_num][0]); +} + #ifdef CONFIG_PM static int bf5xx_i2s_suspend(struct platform_device *dev, struct snd_soc_dai *dai) @@ -263,15 +281,16 @@ struct snd_soc_dai bf5xx_i2s_dai = { .id = 0, .type = SND_SOC_DAI_I2S, .probe = bf5xx_i2s_probe, + .remove = bf5xx_i2s_remove, .suspend = bf5xx_i2s_suspend, .resume = bf5xx_i2s_resume, .playback = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h index 4c163454bbf8..fcadcc081f7f 100644 --- a/sound/soc/blackfin/bf5xx-sport.h +++ b/sound/soc/blackfin/bf5xx-sport.h @@ -123,6 +123,8 @@ struct sport_device { int rx_pos; unsigned int tx_buffer_size; unsigned int rx_buffer_size; + int tx_delay_pos; + int once; #endif void *private_data; }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e0b9869df0f1..4975d8573e4f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -3,9 +3,11 @@ config SND_SOC_ALL_CODECS depends on I2C select SPI select SPI_MASTER + select SND_SOC_AD73311 select SND_SOC_AK4535 select SND_SOC_CS4270 select SND_SOC_SSM2602 + select SND_SOC_TLV320AIC23 select SND_SOC_TLV320AIC26 select SND_SOC_TLV320AIC3X select SND_SOC_UDA1380 @@ -34,6 +36,9 @@ config SND_SOC_AC97_CODEC config SND_SOC_AD1980 tristate +config SND_SOC_AD73311 + tristate + config SND_SOC_AK4535 tristate @@ -58,9 +63,13 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_SSM2602 tristate +config SND_SOC_TLV320AIC23 + tristate + depends on I2C + config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" - depends on SND_SOC && SPI + depends on SPI config SND_SOC_TLV320AIC3X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f977978a3409..90f0a585fc70 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,8 +1,10 @@ snd-soc-ac97-objs := ac97.o snd-soc-ad1980-objs := ad1980.o +snd-soc-ad73311-objs := ad73311.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-uda1380-objs := uda1380.o @@ -20,9 +22,11 @@ snd-soc-wm9713-objs := wm9713.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o +obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 61fd96ca7bc7..bd1ebdc6c86c 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -2,8 +2,7 @@ * ac97.c -- ALSA Soc AC97 codec support * * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@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 diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 4e09c1f2c063..1397b8e06c0b 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -13,7 +13,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h> diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c new file mode 100644 index 000000000000..37af8607b00a --- /dev/null +++ b/sound/soc/codecs/ad73311.c @@ -0,0 +1,107 @@ +/* + * ad73311.c -- ALSA Soc AD73311 codec support + * + * Copyright: Analog Device Inc. + * Author: Cliff Cai <cliff.cai@analog.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Revision history + * 25th Sep 2008 Initial version. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "ad73311.h" + +struct snd_soc_dai ad73311_dai = { + .name = "AD73311", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, +}; +EXPORT_SYMBOL_GPL(ad73311_dai); + +static int ad73311_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + mutex_init(&codec->mutex); + codec->name = "AD73311"; + codec->owner = THIS_MODULE; + codec->dai = &ad73311_dai; + codec->num_dai = 1; + socdev->codec = codec; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ad73311: failed to create pcms\n"); + goto pcm_err; + } + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ad73311: failed to register card\n"); + goto register_err; + } + + return ret; + +register_err: + snd_soc_free_pcms(socdev); +pcm_err: + kfree(socdev->codec); + socdev->codec = NULL; + return ret; +} + +static int ad73311_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec == NULL) + return 0; + snd_soc_free_pcms(socdev); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ad73311 = { + .probe = ad73311_soc_probe, + .remove = ad73311_soc_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311); + +MODULE_DESCRIPTION("ASoC ad73311 driver"); +MODULE_AUTHOR("Cliff Cai "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad73311.h b/sound/soc/codecs/ad73311.h new file mode 100644 index 000000000000..507ce0c30edf --- /dev/null +++ b/sound/soc/codecs/ad73311.h @@ -0,0 +1,90 @@ +/* + * File: sound/soc/codec/ad73311.h + * Based on: + * Author: Cliff Cai <cliff.cai@analog.com> + * + * Created: Thur Sep 25, 2008 + * Description: definitions for AD73311 registers + * + * + * Modified: + * Copyright 2006 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef __AD73311_H__ +#define __AD73311_H__ + +#define AD_CONTROL 0x8000 +#define AD_DATA 0x0000 +#define AD_READ 0x4000 +#define AD_WRITE 0x0000 + +/* Control register A */ +#define CTRL_REG_A (0 << 8) + +#define REGA_MODE_PRO 0x00 +#define REGA_MODE_DATA 0x01 +#define REGA_MODE_MIXED 0x03 +#define REGA_DLB 0x04 +#define REGA_SLB 0x08 +#define REGA_DEVC(x) ((x & 0x7) << 4) +#define REGA_RESET 0x80 + +/* Control register B */ +#define CTRL_REG_B (1 << 8) + +#define REGB_DIRATE(x) (x & 0x3) +#define REGB_SCDIV(x) ((x & 0x3) << 2) +#define REGB_MCDIV(x) ((x & 0x7) << 4) +#define REGB_CEE (1 << 7) + +/* Control register C */ +#define CTRL_REG_C (2 << 8) + +#define REGC_PUDEV (1 << 0) +#define REGC_PUADC (1 << 3) +#define REGC_PUDAC (1 << 4) +#define REGC_PUREF (1 << 5) +#define REGC_REFUSE (1 << 6) + +/* Control register D */ +#define CTRL_REG_D (3 << 8) + +#define REGD_IGS(x) (x & 0x7) +#define REGD_RMOD (1 << 3) +#define REGD_OGS(x) ((x & 0x7) << 4) +#define REGD_MUTE (x << 7) + +/* Control register E */ +#define CTRL_REG_E (4 << 8) + +#define REGE_DA(x) (x & 0x1f) +#define REGE_IBYP (1 << 5) + +/* Control register F */ +#define CTRL_REG_F (5 << 8) + +#define REGF_SEEN (1 << 5) +#define REGF_INV (1 << 6) +#define REGF_ALB (1 << 7) + +extern struct snd_soc_dai ad73311_dai; +extern struct snd_soc_codec_device soc_codec_dev_ad73311; +#endif diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 088cf9927720..2a89b5888e11 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -28,7 +28,6 @@ #include "ak4535.h" -#define AUDIO_NAME "ak4535" #define AK4535_VERSION "0.3" struct snd_soc_codec_device soc_codec_dev_ak4535; diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 940ce1c3522e..44ef0dacd564 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -42,7 +42,6 @@ #include "ssm2602.h" -#define AUDIO_NAME "ssm2602" #define SSM2602_VERSION "0.1" struct snd_soc_codec_device soc_codec_dev_ssm2602; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c new file mode 100644 index 000000000000..bac7815e00fb --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.c @@ -0,0 +1,714 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks@mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd., + * + * Based on sound/soc/codecs/wm8731.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. + * + * Notes: + * The AIC23 is a driver for a low power stereo audio + * codec tlv320aic23 + * + * The machine layer should disable unsupported inputs/outputs by + * snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/initval.h> + +#include "tlv320aic23.h" + +#define AIC23_VERSION "0.1" + +struct tlv320aic23_srate_reg_info { + u32 sample_rate; + u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ + u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ +}; + +/* + * AIC23 register cache + */ +static const u16 tlv320aic23_reg[] = { + 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */ + 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */ + 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */ + 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; + + /* 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 < 0 || reg > 9) && (reg != 15)) { + printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg); + return -1; + } + + data = (reg << 1) | (value >> 8 & 0x01); + + tlv320aic23_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, data, + (value & 0xff)) == 0) + return 0; + + printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__, + value, reg); + + return -EIO; +} + +static const char *rec_src_text[] = { "Line", "Mic" }; +static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; + +static const struct soc_enum rec_src_enum = + SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); + +static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls = +SOC_DAPM_ENUM("Input Select", rec_src_enum); + +static const struct soc_enum tlv320aic23_rec_src = + SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); +static const struct soc_enum tlv320aic23_deemph = + SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text); + +static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0); +static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(sidetone_vol_tlv, -1800, 300, 0); + +static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u16 val, reg; + + val = (ucontrol->value.integer.value[0] & 0x07); + + /* linear conversion to userspace + * 000 = -6db + * 001 = -9db + * 010 = -12db + * 011 = -18db (Min) + * 100 = 0db (Max) + */ + val = (val >= 4) ? 4 : (3 - val); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0); + tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); + + return 0; +} + +static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u16 val; + + val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0); + val = val >> 6; + val = (val >= 4) ? 4 : (3 - val); + ucontrol->value.integer.value[0] = val; + return 0; + +} + +#define SOC_TLV320AIC23_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, .get = snd_soc_tlv320aic23_get_volsw,\ + .put = snd_soc_tlv320aic23_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + +static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { + SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL, + TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv), + SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1), + SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL, + TLV320AIC23_RINVOL, 7, 1, 0), + SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL, + TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv), + SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1), + SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0), + SOC_TLV320AIC23_SINGLE_TLV("Sidetone Volume", TLV320AIC23_ANLG, + 6, 4, 0, sidetone_vol_tlv), + SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph), +}; + +/* add non dapm controls */ +static int tlv320aic23_add_controls(struct snd_soc_codec *codec) +{ + + int err, i; + + for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&tlv320aic23_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; + +} + +/* PGA Mixer controls for Line and Mic switch */ +static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0), + SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1), + SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0, + &tlv320aic23_rec_src_mux_controls), + SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1, + &tlv320aic23_output_mixer_controls[0], + ARRAY_SIZE(tlv320aic23_output_mixer_controls)), + SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0), + + SND_SOC_DAPM_OUTPUT("LHPOUT"), + SND_SOC_DAPM_OUTPUT("RHPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + + SND_SOC_DAPM_INPUT("LLINEIN"), + SND_SOC_DAPM_INPUT("RLINEIN"), + + SND_SOC_DAPM_INPUT("MICIN"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Output Mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "Playback Switch", "DAC"}, + {"Output Mixer", "Mic Sidetone Switch", "Mic Input"}, + + /* Outputs */ + {"RHPOUT", NULL, "Output Mixer"}, + {"LHPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, + + /* Inputs */ + {"Line Input", "NULL", "LLINEIN"}, + {"Line Input", "NULL", "RLINEIN"}, + + {"Mic Input", "NULL", "MICIN"}, + + /* input mux */ + {"Capture Source", "Line", "Line Input"}, + {"Capture Source", "Mic", "Mic Input"}, + {"ADC", NULL, "Capture Source"}, + +}; + +/* tlv320aic23 related */ +static const struct tlv320aic23_srate_reg_info srate_reg_info[] = { + {4000, 0x06, 1}, /* 4000 */ + {8000, 0x06, 0}, /* 8000 */ + {16000, 0x0C, 1}, /* 16000 */ + {22050, 0x11, 1}, /* 22050 */ + {24000, 0x00, 1}, /* 24000 */ + {32000, 0x0C, 0}, /* 32000 */ + {44100, 0x11, 0}, /* 44100 */ + {48000, 0x00, 0}, /* 48000 */ + {88200, 0x1F, 0}, /* 88200 */ + {96000, 0x0E, 0}, /* 96000 */ +}; + +static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int tlv320aic23_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_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 iface_reg, data; + u8 count = 0; + + iface_reg = + tlv320aic23_read_reg_cache(codec, + TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + + /* Search for the right sample rate */ + /* Verify what happens if the rate is not supported + * now it goes to 96Khz */ + while ((srate_reg_info[count].sample_rate != params_rate(params)) && + (count < ARRAY_SIZE(srate_reg_info))) { + count++; + } + + data = (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) | + (srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) | + TLV320AIC23_USB_CLK_ON; + + tlv320aic23_write(codec, TLV320AIC23_SRATE, data); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_reg |= (0x01 << 2); + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_reg |= (0x02 << 2); + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_reg |= (0x03 << 2); + break; + } + tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + + return 0; +} + +static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + + /* set active */ + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001); + + return 0; +} + +static void tlv320aic23_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + + /* deactivate */ + if (!codec->active) { + udelay(50); + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + } +} + +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); + if (mute) + reg |= TLV320AIC23_DACM_MUTE; + + else + reg &= ~TLV320AIC23_DACM_MUTE; + + tlv320aic23_write(codec, TLV320AIC23_DIGT, reg); + + return 0; +} + +static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface_reg; + + iface_reg = + tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface_reg |= TLV320AIC23_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface_reg |= TLV320AIC23_FOR_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg |= TLV320AIC23_FOR_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg |= TLV320AIC23_FOR_LJUST; + break; + default: + return -EINVAL; + + } + + tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + + return 0; +} + +static int tlv320aic23_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; + + switch (freq) { + case 12000000: + return 0; + } + return -EINVAL; +} + +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; + + switch (level) { + case SND_SOC_BIAS_ON: + /* vref/mid, osc on, dac unmute */ + tlv320aic23_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 | 0x0040); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +#define AIC23_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai tlv320aic23_dai = { + .name = "tlv320aic23", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .ops = { + .prepare = tlv320aic23_pcm_prepare, + .hw_params = tlv320aic23_hw_params, + .shutdown = tlv320aic23_shutdown, + }, + .dai_ops = { + .digital_mute = tlv320aic23_mute, + .set_fmt = tlv320aic23_set_dai_fmt, + .set_sysclk = tlv320aic23_set_dai_sysclk, + } +}; +EXPORT_SYMBOL_GPL(tlv320aic23_dai); + +static int tlv320aic23_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int tlv320aic23_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u16 reg; + + /* Sync reg_cache with the hardware */ + for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) { + u16 val = tlv320aic23_read_reg_cache(codec, reg); + tlv320aic23_write(codec, reg, val); + } + + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + tlv320aic23_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +/* + * initialise the AIC23 driver + * register the mixer and dsp interfaces with the kernel + */ +static int tlv320aic23_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + u16 reg; + + codec->name = "tlv320aic23"; + codec->owner = THIS_MODULE; + codec->read = tlv320aic23_read_reg_cache; + codec->write = tlv320aic23_write; + codec->set_bias_level = tlv320aic23_set_bias_level; + codec->dai = &tlv320aic23_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(tlv320aic23_reg); + codec->reg_cache = + kmemdup(tlv320aic23_reg, sizeof(tlv320aic23_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* Reset codec */ + tlv320aic23_write(codec, TLV320AIC23_RESET, 0); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + tlv320aic23_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)); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL); + tlv320aic23_write(codec, TLV320AIC23_RINVOL, + (reg & (~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)); + + /* 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); + + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); + + tlv320aic23_add_controls(codec); + tlv320aic23_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} +static struct snd_soc_device *tlv320aic23_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * If the i2c layer weren't so broken, we could pass this kind of data + * around + */ +static int tlv320aic23_codec_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct snd_soc_device *socdev = tlv320aic23_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = tlv320aic23_init(socdev); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to initialise AIC23\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} +static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c) +{ + put_device(&i2c->dev); + return 0; +} + +static const struct i2c_device_id tlv320aic23_id[] = { + {"tlv320aic23", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, tlv320aic23_id); + +static struct i2c_driver tlv320aic23_i2c_driver = { + .driver = { + .name = "tlv320aic23", + }, + .probe = tlv320aic23_codec_probe, + .remove = __exit_p(tlv320aic23_i2c_remove), + .id_table = tlv320aic23_id, +}; + +#endif + +static int tlv320aic23_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + tlv320aic23_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data; + codec->hw_read = NULL; + ret = i2c_add_driver(&tlv320aic23_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); +#endif + return ret; +} + +static int tlv320aic23_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&tlv320aic23_i2c_driver); +#endif + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} +struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = { + .probe = tlv320aic23_probe, + .remove = tlv320aic23_remove, + .suspend = tlv320aic23_suspend, + .resume = tlv320aic23_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23); + +MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); +MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h new file mode 100644 index 000000000000..79d1faf8e570 --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.h @@ -0,0 +1,122 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks@mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd + * + * 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 _TLV320AIC23_H +#define _TLV320AIC23_H + +/* Codec TLV320AIC23 */ +#define TLV320AIC23_LINVOL 0x00 +#define TLV320AIC23_RINVOL 0x01 +#define TLV320AIC23_LCHNVOL 0x02 +#define TLV320AIC23_RCHNVOL 0x03 +#define TLV320AIC23_ANLG 0x04 +#define TLV320AIC23_DIGT 0x05 +#define TLV320AIC23_PWR 0x06 +#define TLV320AIC23_DIGT_FMT 0x07 +#define TLV320AIC23_SRATE 0x08 +#define TLV320AIC23_ACTIVE 0x09 +#define TLV320AIC23_RESET 0x0F + +/* Left (right) line input volume control register */ +#define TLV320AIC23_LRS_ENABLED 0x0100 +#define TLV320AIC23_LIM_MUTED 0x0080 +#define TLV320AIC23_LIV_DEFAULT 0x0017 +#define TLV320AIC23_LIV_MAX 0x001f +#define TLV320AIC23_LIV_MIN 0x0000 + +/* Left (right) channel headphone volume control register */ +#define TLV320AIC23_LZC_ON 0x0080 +#define TLV320AIC23_LHV_DEFAULT 0x0079 +#define TLV320AIC23_LHV_MAX 0x007f +#define TLV320AIC23_LHV_MIN 0x0000 + +/* Analog audio path control register */ +#define TLV320AIC23_STA_REG(x) ((x)<<6) +#define TLV320AIC23_STE_ENABLED 0x0020 +#define TLV320AIC23_DAC_SELECTED 0x0010 +#define TLV320AIC23_BYPASS_ON 0x0008 +#define TLV320AIC23_INSEL_MIC 0x0004 +#define TLV320AIC23_MICM_MUTED 0x0002 +#define TLV320AIC23_MICB_20DB 0x0001 + +/* Digital audio path control register */ +#define TLV320AIC23_DACM_MUTE 0x0008 +#define TLV320AIC23_DEEMP_32K 0x0002 +#define TLV320AIC23_DEEMP_44K 0x0004 +#define TLV320AIC23_DEEMP_48K 0x0006 +#define TLV320AIC23_ADCHP_ON 0x0001 + +/* Power control down register */ +#define TLV320AIC23_DEVICE_PWR_OFF 0x0080 +#define TLV320AIC23_CLK_OFF 0x0040 +#define TLV320AIC23_OSC_OFF 0x0020 +#define TLV320AIC23_OUT_OFF 0x0010 +#define TLV320AIC23_DAC_OFF 0x0008 +#define TLV320AIC23_ADC_OFF 0x0004 +#define TLV320AIC23_MIC_OFF 0x0002 +#define TLV320AIC23_LINE_OFF 0x0001 + +/* Digital audio interface register */ +#define TLV320AIC23_MS_MASTER 0x0040 +#define TLV320AIC23_LRSWAP_ON 0x0020 +#define TLV320AIC23_LRP_ON 0x0010 +#define TLV320AIC23_IWL_16 0x0000 +#define TLV320AIC23_IWL_20 0x0004 +#define TLV320AIC23_IWL_24 0x0008 +#define TLV320AIC23_IWL_32 0x000C +#define TLV320AIC23_FOR_I2S 0x0002 +#define TLV320AIC23_FOR_DSP 0x0003 +#define TLV320AIC23_FOR_LJUST 0x0001 + +/* Sample rate control register */ +#define TLV320AIC23_CLKOUT_HALF 0x0080 +#define TLV320AIC23_CLKIN_HALF 0x0040 +#define TLV320AIC23_BOSR_384fs 0x0002 /* BOSR_272fs in USB mode */ +#define TLV320AIC23_USB_CLK_ON 0x0001 +#define TLV320AIC23_SR_MASK 0xf +#define TLV320AIC23_CLKOUT_SHIFT 7 +#define TLV320AIC23_CLKIN_SHIFT 6 +#define TLV320AIC23_SR_SHIFT 2 +#define TLV320AIC23_BOSR_SHIFT 1 + +/* Digital interface register */ +#define TLV320AIC23_ACT_ON 0x0001 + +/* + * AUDIO related MACROS + */ + +#define TLV320AIC23_DEFAULT_OUT_VOL 0x70 +#define TLV320AIC23_DEFAULT_IN_VOLUME 0x10 + +#define TLV320AIC23_OUT_VOL_MIN TLV320AIC23_LHV_MIN +#define TLV320AIC23_OUT_VOL_MAX TLV320AIC23_LHV_MAX +#define TLV320AIC23_OUT_VO_RANGE (TLV320AIC23_OUT_VOL_MAX - \ + TLV320AIC23_OUT_VOL_MIN) +#define TLV320AIC23_OUT_VOL_MASK TLV320AIC23_OUT_VOL_MAX + +#define TLV320AIC23_IN_VOL_MIN TLV320AIC23_LIV_MIN +#define TLV320AIC23_IN_VOL_MAX TLV320AIC23_LIV_MAX +#define TLV320AIC23_IN_VOL_RANGE (TLV320AIC23_IN_VOL_MAX - \ + TLV320AIC23_IN_VOL_MIN) +#define TLV320AIC23_IN_VOL_MASK TLV320AIC23_IN_VOL_MAX + +#define TLV320AIC23_SIDETONE_MASK 0x1c0 +#define TLV320AIC23_SIDETONE_0 0x100 +#define TLV320AIC23_SIDETONE_6 0x000 +#define TLV320AIC23_SIDETONE_9 0x040 +#define TLV320AIC23_SIDETONE_12 0x080 +#define TLV320AIC23_SIDETONE_18 0x0c0 + +extern struct snd_soc_dai tlv320aic23_dai; +extern struct snd_soc_codec_device soc_codec_dev_tlv320aic23; + +#endif /* _TLV320AIC23_H */ diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 566a427c928f..05336ed7e493 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -48,7 +48,6 @@ #include "tlv320aic3x.h" -#define AUDIO_NAME "aic3x" #define AIC3X_VERSION "0.2" /* codec private data */ @@ -991,7 +990,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected); SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) struct snd_soc_dai aic3x_dai = { - .name = "aic3x", + .name = "tlv320aic3x", .playback = { .stream_name = "Playback", .channels_min = 1, @@ -1055,7 +1054,7 @@ static int aic3x_init(struct snd_soc_device *socdev) struct aic3x_setup_data *setup = socdev->codec_data; int reg, ret = 0; - codec->name = "aic3x"; + codec->name = "tlv320aic3x"; codec->owner = THIS_MODULE; codec->read = aic3x_read_reg_cache; codec->write = aic3x_write; diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index d206d7f892b6..a69ee72a7af5 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -36,7 +36,6 @@ #include "uda1380.h" #define UDA1380_VERSION "0.6" -#define AUDIO_NAME "uda1380" /* * uda1380 register cache diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 9a37c8d95ed2..d8ca2da8d634 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -3,7 +3,7 @@ * * Copyright 2006 Wolfson Microelectronics PLC. * - * Author: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Author: Liam Girdwood <lrg@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 version 2 as @@ -18,6 +18,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -27,7 +28,6 @@ #include "wm8510.h" -#define AUDIO_NAME "wm8510" #define WM8510_VERSION "0.6" struct snd_soc_codec_device soc_codec_dev_wm8510; @@ -55,6 +55,9 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { 0x0001, }; +#define WM8510_POWER1_BIASEN 0x08 +#define WM8510_POWER1_BUFIOEN 0x10 + /* * read wm8510 register cache */ @@ -224,9 +227,9 @@ SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0), -SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0, - &wm8510_micpga_controls[0], - ARRAY_SIZE(wm8510_micpga_controls)), +SND_SOC_DAPM_MIXER("Mic PGA", WM8510_POWER2, 2, 0, + &wm8510_micpga_controls[0], + ARRAY_SIZE(wm8510_micpga_controls)), SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0, &wm8510_boost_controls[0], ARRAY_SIZE(wm8510_boost_controls)), @@ -526,23 +529,35 @@ static int wm8510_mute(struct snd_soc_dai *dai, int mute) static int wm8510_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + u16 power1 = wm8510_read_reg_cache(codec, WM8510_POWER1) & ~0x3; switch (level) { case SND_SOC_BIAS_ON: - wm8510_write(codec, WM8510_POWER1, 0x1ff); - wm8510_write(codec, WM8510_POWER2, 0x1ff); - wm8510_write(codec, WM8510_POWER3, 0x1ff); - break; case SND_SOC_BIAS_PREPARE: + power1 |= 0x1; /* VMID 50k */ + wm8510_write(codec, WM8510_POWER1, power1); + break; + case SND_SOC_BIAS_STANDBY: + power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN; + + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Initial cap charge at VMID 5k */ + wm8510_write(codec, WM8510_POWER1, power1 | 0x3); + mdelay(100); + } + + power1 |= 0x2; /* VMID 500k */ + wm8510_write(codec, WM8510_POWER1, power1); break; + case SND_SOC_BIAS_OFF: - /* everything off, dac mute, inactive */ - wm8510_write(codec, WM8510_POWER1, 0x0); - wm8510_write(codec, WM8510_POWER2, 0x0); - wm8510_write(codec, WM8510_POWER3, 0x0); + wm8510_write(codec, WM8510_POWER1, 0); + wm8510_write(codec, WM8510_POWER2, 0); + wm8510_write(codec, WM8510_POWER3, 0); break; } + codec->bias_level = level; return 0; } @@ -640,6 +655,7 @@ static int wm8510_init(struct snd_soc_device *socdev) } /* power on device */ + codec->bias_level = SND_SOC_BIAS_OFF; wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); wm8510_add_controls(codec); wm8510_add_widgets(codec); @@ -747,6 +763,62 @@ err_driver: } #endif +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8510_spi_probe(struct spi_device *spi) +{ + struct snd_soc_device *socdev = wm8510_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + codec->control_data = spi; + + ret = wm8510_init(socdev); + if (ret < 0) + dev_err(&spi->dev, "failed to initialise WM8510\n"); + + return ret; +} + +static int __devexit wm8510_spi_remove(struct spi_device *spi) +{ + return 0; +} + +static struct spi_driver wm8510_spi_driver = { + .driver = { + .name = "wm8510", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8510_spi_probe, + .remove = __devexit_p(wm8510_spi_remove), +}; + +static int wm8510_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#endif /* CONFIG_SPI_MASTER */ + static int wm8510_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -772,8 +844,14 @@ static int wm8510_probe(struct platform_device *pdev) codec->hw_write = (hw_write_t)i2c_master_send; ret = wm8510_add_i2c_device(pdev, setup); } -#else - /* Add other interfaces here */ +#endif +#if defined(CONFIG_SPI_MASTER) + if (setup->spi) { + codec->hw_write = (hw_write_t)wm8510_spi_write; + ret = spi_register_driver(&wm8510_spi_driver); + if (ret != 0) + printk(KERN_ERR "can't add spi driver"); + } #endif if (ret != 0) @@ -796,6 +874,9 @@ static int wm8510_remove(struct platform_device *pdev) i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8510_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8510_spi_driver); +#endif kfree(codec); return 0; diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h index c53683960456..bdefcf5c69ff 100644 --- a/sound/soc/codecs/wm8510.h +++ b/sound/soc/codecs/wm8510.h @@ -94,6 +94,7 @@ #define WM8510_MCLKDIV_12 (7 << 5) struct wm8510_setup_data { + int spi; int i2c_bus; unsigned short i2c_address; }; diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index df1ffbe305bf..627ebfb4209b 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -18,7 +18,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> @@ -36,7 +35,6 @@ #include "wm8580.h" -#define AUDIO_NAME "wm8580" #define WM8580_VERSION "0.1" struct pll_state { diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7b64d9a7ff76..7f8a7e36b33e 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -29,7 +29,6 @@ #include "wm8731.h" -#define AUDIO_NAME "wm8731" #define WM8731_VERSION "0.13" struct snd_soc_codec_device soc_codec_dev_wm8731; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 4892e398a598..9b7296ee5b08 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -29,7 +29,6 @@ #include "wm8750.h" -#define AUDIO_NAME "WM8750" #define WM8750_VERSION "0.12" /* codec private data */ diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 8c4df44f3345..d426eaa22185 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -2,8 +2,7 @@ * wm8753.c -- WM8753 ALSA Soc Audio driver * * Copyright 2003 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@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 @@ -40,6 +39,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -51,7 +51,6 @@ #include "wm8753.h" -#define AUDIO_NAME "wm8753" #define WM8753_VERSION "0.16" static int caps_charge = 2000; @@ -1719,6 +1718,63 @@ err_driver: } #endif +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8753_spi_probe(struct spi_device *spi) +{ + struct snd_soc_device *socdev = wm8753_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + codec->control_data = spi; + + ret = wm8753_init(socdev); + if (ret < 0) + dev_err(&spi->dev, "failed to initialise WM8753\n"); + + return ret; +} + +static int __devexit wm8753_spi_remove(struct spi_device *spi) +{ + return 0; +} + +static struct spi_driver wm8753_spi_driver = { + .driver = { + .name = "wm8753", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8753_spi_probe, + .remove = __devexit_p(wm8753_spi_remove), +}; + +static int wm8753_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#endif + + static int wm8753_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -1753,8 +1809,14 @@ static int wm8753_probe(struct platform_device *pdev) codec->hw_write = (hw_write_t)i2c_master_send; ret = wm8753_add_i2c_device(pdev, setup); } -#else - /* Add other interfaces here */ +#endif +#if defined(CONFIG_SPI_MASTER) + if (setup->spi) { + codec->hw_write = (hw_write_t)wm8753_spi_write; + ret = spi_register_driver(&wm8753_spi_driver); + if (ret != 0) + printk(KERN_ERR "can't add spi driver"); + } #endif if (ret != 0) { @@ -1798,6 +1860,9 @@ static int wm8753_remove(struct platform_device *pdev) i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8753_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8753_spi_driver); +#endif kfree(codec->private_data); kfree(codec); diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h index 7defde069f1d..f55704ce931b 100644 --- a/sound/soc/codecs/wm8753.h +++ b/sound/soc/codecs/wm8753.h @@ -2,8 +2,7 @@ * wm8753.h -- audio driver for WM8753 * * Copyright 2003 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@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 @@ -79,6 +78,7 @@ #define WM8753_ADCTL2 0x3f struct wm8753_setup_data { + int spi; int i2c_bus; unsigned short i2c_address; }; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 0b8c6d38b48f..3b326c9b5586 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -18,7 +18,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index a3f54ec4226e..ce40d7877605 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -653,14 +653,14 @@ static const struct snd_kcontrol_new wm8903_snd_controls[] = { /* Input PGAs - No TLV since the scale depends on PGA mode */ SOC_SINGLE("Left Input PGA Switch", WM8903_ANALOGUE_LEFT_INPUT_0, - 7, 1, 0), + 7, 1, 1), SOC_SINGLE("Left Input PGA Volume", WM8903_ANALOGUE_LEFT_INPUT_0, 0, 31, 0), SOC_SINGLE("Left Input PGA Common Mode Switch", WM8903_ANALOGUE_LEFT_INPUT_1, 6, 1, 0), SOC_SINGLE("Right Input PGA Switch", WM8903_ANALOGUE_RIGHT_INPUT_0, - 7, 1, 0), + 7, 1, 1), SOC_SINGLE("Right Input PGA Volume", WM8903_ANALOGUE_RIGHT_INPUT_0, 0, 31, 0), SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1, diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 974a4cd0f3fd..f41a578ddd4f 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -29,7 +29,6 @@ #include "wm8971.h" -#define AUDIO_NAME "wm8971" #define WM8971_VERSION "0.9" #define WM8971_REG_COUNT 43 diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 63410d7b5efb..572d22b0880b 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -30,7 +30,6 @@ #include "wm8990.h" -#define AUDIO_NAME "wm8990" #define WM8990_VERSION "0.2" /* codec private data */ diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 2f1c91b1d556..ffb471e420e2 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -2,8 +2,7 @@ * wm9712.c -- ALSA Soc WM9712 codec support * * Copyright 2006 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@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 diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 441d0580db1f..aba402b3c999 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -2,8 +2,7 @@ * wm9713.c -- ALSA Soc WM9713 codec support * * Copyright 2006 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@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 diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index aea27e70043c..8b7766b998d7 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -13,3 +13,11 @@ config SND_OMAP_SOC_N810 select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. + +config SND_OMAP_SOC_OSK5912 + tristate "SoC Audio support for omap osk5912" + depends on SND_OMAP_SOC && MACH_OMAP_OSK + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC23 + help + Say Y if you want to add support for SoC audio on osk5912. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index d8d8d58075e3..e09d1f297f64 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -7,5 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o # OMAP Machine Support snd-soc-n810-objs := n810.o +snd-soc-osk5912-objs := osk5912.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index d166b6b2a60d..fae3ad36e0bf 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -247,9 +247,9 @@ static int n810_aic33_init(struct snd_soc_codec *codec) int i, err; /* Not connected */ - snd_soc_dapm_disable_pin(codec, "MONO_LOUT"); - snd_soc_dapm_disable_pin(codec, "HPLCOM"); - snd_soc_dapm_disable_pin(codec, "HPRCOM"); + snd_soc_dapm_nc_pin(codec, "MONO_LOUT"); + snd_soc_dapm_nc_pin(codec, "HPLCOM"); + snd_soc_dapm_nc_pin(codec, "HPRCOM"); /* Add N810 specific controls */ for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) { diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 35310e16d7f3..0a063a98a661 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -59,12 +59,7 @@ static struct omap_mcbsp_data mcbsp_data[NUM_LINKS]; * Stream DMA parameters. DMA request line and port address are set runtime * since they are different between OMAP1 and later OMAPs */ -static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2] = { -{ - { .name = "I2S PCM Stereo out", }, - { .name = "I2S PCM Stereo in", }, -}, -}; +static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2]; #if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX) static const int omap1_dma_reqs[][2] = { @@ -84,11 +79,22 @@ static const unsigned long omap1_mcbsp_port[][2] = { static const int omap1_dma_reqs[][2] = {}; static const unsigned long omap1_mcbsp_port[][2] = {}; #endif -#if defined(CONFIG_ARCH_OMAP2420) -static const int omap2420_dma_reqs[][2] = { + +#if defined(CONFIG_ARCH_OMAP24XX) || defined(CONFIG_ARCH_OMAP34XX) +static const int omap24xx_dma_reqs[][2] = { { OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX }, { OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX }, +#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX) + { OMAP24XX_DMA_MCBSP3_TX, OMAP24XX_DMA_MCBSP3_RX }, + { OMAP24XX_DMA_MCBSP4_TX, OMAP24XX_DMA_MCBSP4_RX }, + { OMAP24XX_DMA_MCBSP5_TX, OMAP24XX_DMA_MCBSP5_RX }, +#endif }; +#else +static const int omap24xx_dma_reqs[][2] = {}; +#endif + +#if defined(CONFIG_ARCH_OMAP2420) static const unsigned long omap2420_mcbsp_port[][2] = { { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1, OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 }, @@ -96,10 +102,43 @@ static const unsigned long omap2420_mcbsp_port[][2] = { OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 }, }; #else -static const int omap2420_dma_reqs[][2] = {}; static const unsigned long omap2420_mcbsp_port[][2] = {}; #endif +#if defined(CONFIG_ARCH_OMAP2430) +static const unsigned long omap2430_mcbsp_port[][2] = { + { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, + OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR, + OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DXR, + OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DXR, + OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DXR, + OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DRR }, +}; +#else +static const unsigned long omap2430_mcbsp_port[][2] = {}; +#endif + +#if defined(CONFIG_ARCH_OMAP34XX) +static const unsigned long omap34xx_mcbsp_port[][2] = { + { OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DRR }, +}; +#else +static const unsigned long omap34xx_mcbsp_port[][2] = {}; +#endif + static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -167,14 +206,19 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, dma = omap1_dma_reqs[bus_id][substream->stream]; port = omap1_mcbsp_port[bus_id][substream->stream]; } else if (cpu_is_omap2420()) { - dma = omap2420_dma_reqs[bus_id][substream->stream]; + dma = omap24xx_dma_reqs[bus_id][substream->stream]; port = omap2420_mcbsp_port[bus_id][substream->stream]; + } else if (cpu_is_omap2430()) { + dma = omap24xx_dma_reqs[bus_id][substream->stream]; + port = omap2430_mcbsp_port[bus_id][substream->stream]; + } else if (cpu_is_omap343x()) { + dma = omap24xx_dma_reqs[bus_id][substream->stream]; + port = omap34xx_mcbsp_port[bus_id][substream->stream]; } else { - /* - * TODO: Add support for 2430 and 3430 - */ return -ENODEV; } + omap_mcbsp_dai_dma_params[id][substream->stream].name = + substream->stream ? "Audio Capture" : "Audio Playback"; omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma; omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port; cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream]; @@ -245,6 +289,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); break; + case SND_SOC_DAIFMT_DSP_A: + /* 0-bit data delay */ + regs->rcr2 |= RDATDLY(0); + regs->xcr2 |= XDATDLY(0); + break; default: /* Unsupported data format */ return -EINVAL; @@ -310,7 +359,7 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data, int clk_id) { int sel_bit; - u16 reg; + u16 reg, reg_devconf1 = OMAP243X_CONTROL_DEVCONF1; if (cpu_class_is_omap1()) { /* OMAP1's can use only external source clock */ @@ -320,6 +369,12 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data, return 0; } + if (cpu_is_omap2420() && mcbsp_data->bus_id > 1) + return -EINVAL; + + if (cpu_is_omap343x()) + reg_devconf1 = OMAP343X_CONTROL_DEVCONF1; + switch (mcbsp_data->bus_id) { case 0: reg = OMAP2_CONTROL_DEVCONF0; @@ -329,20 +384,26 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data, reg = OMAP2_CONTROL_DEVCONF0; sel_bit = 6; break; - /* TODO: Support for ports 3 - 5 in OMAP2430 and OMAP34xx */ + case 2: + reg = reg_devconf1; + sel_bit = 0; + break; + case 3: + reg = reg_devconf1; + sel_bit = 2; + break; + case 4: + reg = reg_devconf1; + sel_bit = 4; + break; default: return -EINVAL; } - if (cpu_class_is_omap2()) { - if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK) { - omap_ctrl_writel(omap_ctrl_readl(reg) & - ~(1 << sel_bit), reg); - } else { - omap_ctrl_writel(omap_ctrl_readl(reg) | - (1 << sel_bit), reg); - } - } + if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK) + omap_ctrl_writel(omap_ctrl_readl(reg) & ~(1 << sel_bit), reg); + else + omap_ctrl_writel(omap_ctrl_readl(reg) | (1 << sel_bit), reg); return 0; } @@ -376,37 +437,49 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return err; } -struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS] = { -{ - .name = "omap-mcbsp-dai", - .id = 0, - .type = SND_SOC_DAI_I2S, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = OMAP_MCBSP_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = OMAP_MCBSP_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = { - .startup = omap_mcbsp_dai_startup, - .shutdown = omap_mcbsp_dai_shutdown, - .trigger = omap_mcbsp_dai_trigger, - .hw_params = omap_mcbsp_dai_hw_params, - }, - .dai_ops = { - .set_fmt = omap_mcbsp_dai_set_dai_fmt, - .set_clkdiv = omap_mcbsp_dai_set_clkdiv, - .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, - }, - .private_data = &mcbsp_data[0].bus_id, -}, +#define OMAP_MCBSP_DAI_BUILDER(link_id) \ +{ \ + .name = "omap-mcbsp-dai-(link_id)", \ + .id = (link_id), \ + .type = SND_SOC_DAI_I2S, \ + .playback = { \ + .channels_min = 2, \ + .channels_max = 2, \ + .rates = OMAP_MCBSP_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ + .capture = { \ + .channels_min = 2, \ + .channels_max = 2, \ + .rates = OMAP_MCBSP_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ + .ops = { \ + .startup = omap_mcbsp_dai_startup, \ + .shutdown = omap_mcbsp_dai_shutdown, \ + .trigger = omap_mcbsp_dai_trigger, \ + .hw_params = omap_mcbsp_dai_hw_params, \ + }, \ + .dai_ops = { \ + .set_fmt = omap_mcbsp_dai_set_dai_fmt, \ + .set_clkdiv = omap_mcbsp_dai_set_clkdiv, \ + .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, \ + }, \ + .private_data = &mcbsp_data[(link_id)].bus_id, \ +} + +struct snd_soc_dai omap_mcbsp_dai[] = { + OMAP_MCBSP_DAI_BUILDER(0), + OMAP_MCBSP_DAI_BUILDER(1), +#if NUM_LINKS >= 3 + OMAP_MCBSP_DAI_BUILDER(2), +#endif +#if NUM_LINKS == 5 + OMAP_MCBSP_DAI_BUILDER(3), + OMAP_MCBSP_DAI_BUILDER(4), +#endif }; + EXPORT_SYMBOL_GPL(omap_mcbsp_dai); MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>"); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index ed8afb550671..df7ad13ba73d 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -38,11 +38,17 @@ enum omap_mcbsp_div { OMAP_MCBSP_CLKGDV, /* Sample rate generator divider */ }; -/* - * REVISIT: Preparation for the ASoC v2. Let the number of available links to - * be same than number of McBSP ports found in OMAP(s) we are compiling for. - */ -#define NUM_LINKS 1 +#if defined(CONFIG_ARCH_OMAP2420) +#define NUM_LINKS 2 +#endif +#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX) +#undef NUM_LINKS +#define NUM_LINKS 3 +#endif +#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX) +#undef NUM_LINKS +#define NUM_LINKS 5 +#endif extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS]; diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 690bfeaec4a0..e9084fdd2082 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -97,7 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, prtd->dma_data = dma_data; err = omap_request_dma(dma_data->dma_req, dma_data->name, omap_pcm_dma_irq, substream, &prtd->dma_ch); - if (!cpu_is_omap1510()) { + if (!err & !cpu_is_omap1510()) { /* * Link channel with itself so DMA doesn't need any * reprogramming while looping the buffer @@ -147,12 +147,14 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_or_dst_synch = OMAP_DMA_DST_SYNC; dma_params.src_start = runtime->dma_addr; dma_params.dst_start = dma_data->port_addr; + dma_params.dst_port = OMAP_DMA_PORT_MPUI; } else { dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT; dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC; dma_params.src_or_dst_synch = OMAP_DMA_SRC_SYNC; dma_params.src_start = dma_data->port_addr; dma_params.dst_start = runtime->dma_addr; + dma_params.src_port = OMAP_DMA_PORT_MPUI; } /* * Set DMA transfer frame size equal to ALSA period size and frame diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c new file mode 100644 index 000000000000..0fe733796898 --- /dev/null +++ b/sound/soc/omap/osk5912.c @@ -0,0 +1,232 @@ +/* + * osk5912.c -- SoC audio for OSK 5912 + * + * Copyright (C) 2008 Mistral Solutions + * + * Contact: Arun KS <arunks@mistralsolutions.com> + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <linux/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/tlv320aic23.h" + +#define CODEC_CLOCK 12000000 + +static struct clk *tlv320aic23_mclk; + +static int osk_startup(struct snd_pcm_substream *substream) +{ + return clk_enable(tlv320aic23_mclk); +} + +static void osk_shutdown(struct snd_pcm_substream *substream) +{ + clk_disable(tlv320aic23_mclk); +} + +static int osk_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err; + + /* Set codec DAI configuration */ + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return err; + } + + /* Set cpu DAI configuration */ + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return err; + } + + /* Set the codec system clock for DAC and ADC */ + err = + snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); + + if (err < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return err; + } + + return err; +} + +static struct snd_soc_ops osk_ops = { + .startup = osk_startup, + .hw_params = osk_hw_params, + .shutdown = osk_shutdown, +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LHPOUT"}, + {"Headphone Jack", NULL, "RHPOUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic Jack"}, +}; + +static int osk_tlv320aic23_init(struct snd_soc_codec *codec) +{ + + /* Add osk5912 specific widgets */ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* Set up osk5912 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link osk_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &tlv320aic23_dai, + .init = osk_tlv320aic23_init, + .ops = &osk_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_osk = { + .name = "OSK5912", + .dai_link = &osk_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device osk_snd_devdata = { + .machine = &snd_soc_machine_osk, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *osk_snd_device; + +static int __init osk_soc_init(void) +{ + int err; + u32 curRate; + struct device *dev; + + if (!(machine_is_omap_osk())) + return -ENODEV; + + osk_snd_device = platform_device_alloc("soc-audio", -1); + if (!osk_snd_device) + return -ENOMEM; + + platform_set_drvdata(osk_snd_device, &osk_snd_devdata); + osk_snd_devdata.dev = &osk_snd_device->dev; + *(unsigned int *)osk_dai.cpu_dai->private_data = 0; /* McBSP1 */ + err = platform_device_add(osk_snd_device); + if (err) + goto err1; + + dev = &osk_snd_device->dev; + + tlv320aic23_mclk = clk_get(dev, "mclk"); + if (IS_ERR(tlv320aic23_mclk)) { + printk(KERN_ERR "Could not get mclk clock\n"); + return -ENODEV; + } + + if (clk_get_usecount(tlv320aic23_mclk) > 0) { + /* MCLK is already in use */ + printk(KERN_WARNING + "MCLK in use at %d Hz. We change it to %d Hz\n", + (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK); + } + + /* + * Configure 12 MHz output on MCLK. + */ + curRate = (uint) clk_get_rate(tlv320aic23_mclk); + if (curRate != CODEC_CLOCK) { + if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) { + printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n"); + err = -ECANCELED; + goto err1; + } + } + + printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n", + (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK, + clk_get_usecount(tlv320aic23_mclk)); + + return 0; +err1: + clk_put(tlv320aic23_mclk); + platform_device_del(osk_snd_device); + platform_device_put(osk_snd_device); + + return err; + +} + +static void __exit osk_soc_exit(void) +{ + platform_device_unregister(osk_snd_device); +} + +module_init(osk_soc_init); +module_exit(osk_soc_exit); + +MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>"); +MODULE_DESCRIPTION("ALSA SoC OSK 5912"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 1a8373de7f3a..2718eaf7895f 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Authors: Liam Girdwood <lrg@slimlogic.co.uk> * Richard Purdie <richard@openedhand.com> * * This program is free software; you can redistribute it and/or modify it @@ -281,8 +281,8 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_disable_pin(codec, "LLINEIN"); - snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_nc_pin(codec, "LLINEIN"); + snd_soc_dapm_nc_pin(codec, "RLINEIN"); /* Add corgi specific controls */ for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) { diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index d9c3f7b28be2..e6ff6929ab4b 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -9,7 +9,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Authors: Liam Girdwood <lrg@slimlogic.co.uk> * Richard Purdie <richard@openedhand.com> * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index f84f7d8db09a..4d9930c52789 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Authors: Liam Girdwood <lrg@slimlogic.co.uk> * Richard Purdie <richard@openedhand.com> * * This program is free software; you can redistribute it and/or modify it @@ -242,8 +242,8 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_disable_pin(codec, "LLINEIN"); - snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_nc_pin(codec, "LLINEIN"); + snd_soc_dapm_nc_pin(codec, "RLINEIN"); snd_soc_dapm_enable_pin(codec, "MICIN"); /* Add poodle specific controls */ diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 2fb58298513b..e758034db5c3 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -3,7 +3,7 @@ * * Copyright 2005 Wolfson Microelectronics PLC. * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * lrg@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 @@ -405,6 +405,6 @@ module_init(pxa2xx_i2s_init); module_exit(pxa2xx_i2s_exit); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk"); MODULE_DESCRIPTION("pxa2xx I2S SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 9a70b00fc30e..d307b6757e95 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Authors: Liam Girdwood <lrg@slimlogic.co.uk> * Richard Purdie <richard@openedhand.com> * * This program is free software; you can redistribute it and/or modify it @@ -281,13 +281,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) int i, err; /* NC codec pins */ - snd_soc_dapm_disable_pin(codec, "RINPUT1"); - snd_soc_dapm_disable_pin(codec, "LINPUT2"); - snd_soc_dapm_disable_pin(codec, "RINPUT2"); - snd_soc_dapm_disable_pin(codec, "LINPUT3"); - snd_soc_dapm_disable_pin(codec, "RINPUT3"); - snd_soc_dapm_disable_pin(codec, "OUT3"); - snd_soc_dapm_disable_pin(codec, "MONO1"); + snd_soc_dapm_nc_pin(codec, "RINPUT1"); + snd_soc_dapm_nc_pin(codec, "LINPUT2"); + snd_soc_dapm_nc_pin(codec, "RINPUT2"); + snd_soc_dapm_nc_pin(codec, "LINPUT3"); + snd_soc_dapm_nc_pin(codec, "RINPUT3"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "MONO1"); /* Add spitz specific controls */ for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) { diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 2baaa750f123..afefe41b8c46 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Authors: Liam Girdwood <lrg@slimlogic.co.uk> * Richard Purdie <richard@openedhand.com> * * This program is free software; you can redistribute it and/or modify it @@ -190,8 +190,8 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_disable_pin(codec, "OUT3"); - snd_soc_dapm_disable_pin(codec, "MONOOUT"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "MONOOUT"); /* add tosa specific controls */ for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) { diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 73a50e93a9a2..87ddfefcc2fb 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -511,21 +511,20 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) DBG("Entered %s\n", __func__); /* set up NC codec pins */ - snd_soc_dapm_disable_pin(codec, "LOUT2"); - snd_soc_dapm_disable_pin(codec, "ROUT2"); - snd_soc_dapm_disable_pin(codec, "OUT3"); - snd_soc_dapm_disable_pin(codec, "OUT4"); - snd_soc_dapm_disable_pin(codec, "LINE1"); - snd_soc_dapm_disable_pin(codec, "LINE2"); - - - /* set endpoints to default mode */ - set_scenario_endpoints(codec, NEO_AUDIO_OFF); + snd_soc_dapm_nc_pin(codec, "LOUT2"); + snd_soc_dapm_nc_pin(codec, "ROUT2"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "OUT4"); + snd_soc_dapm_nc_pin(codec, "LINE1"); + snd_soc_dapm_nc_pin(codec, "LINE2"); /* Add neo1973 specific widgets */ snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, ARRAY_SIZE(wm8753_dapm_widgets)); + /* set endpoints to default mode */ + set_scenario_endpoints(codec, NEO_AUDIO_OFF); + /* add neo1973 specific controls */ for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) { err = snd_ctl_add(codec->card, @@ -603,6 +602,8 @@ static int lm4857_i2c_probe(struct i2c_client *client, { DBG("Entered %s\n", __func__); + i2c = client; + lm4857_write_regs(); return 0; } @@ -611,6 +612,8 @@ static int lm4857_i2c_remove(struct i2c_client *client) { DBG("Entered %s\n", __func__); + i2c = NULL; + return 0; } @@ -650,7 +653,7 @@ static void lm4857_shutdown(struct i2c_client *dev) } static const struct i2c_device_id lm4857_i2c_id[] = { - { "neo1973_lm4857", 0 } + { "neo1973_lm4857", 0 }, { } }; @@ -668,48 +671,6 @@ static struct i2c_driver lm4857_i2c_driver = { }; static struct platform_device *neo1973_snd_device; -static struct i2c_client *lm4857_client; - -static int __init neo1973_add_lm4857_device(struct platform_device *pdev, - int i2c_bus, - unsigned short i2c_address) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - int ret; - - ret = i2c_add_driver(&lm4857_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add lm4857 driver\n"); - return ret; - } - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = i2c_address; - strlcpy(info.type, "neo1973_lm4857", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "can't get i2c adapter %d\n", i2c_bus); - goto err_driver; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - dev_err(&pdev->dev, "can't add lm4857 device at 0x%x\n", - (unsigned int)info.addr); - goto err_driver; - } - - lm4857_client = client; - return 0; - -err_driver: - i2c_del_driver(&lm4857_i2c_driver); - return -ENODEV; -} static int __init neo1973_init(void) { @@ -736,8 +697,8 @@ static int __init neo1973_init(void) return ret; } - ret = neo1973_add_lm4857_device(neo1973_snd_device, - neo1973_wm8753_setup, 0x7C); + ret = i2c_add_driver(&lm4857_i2c_driver); + if (ret != 0) platform_device_unregister(neo1973_snd_device); @@ -748,7 +709,6 @@ static void __exit neo1973_exit(void) { DBG("Entered %s\n", __func__); - i2c_unregister_device(lm4857_client); i2c_del_driver(&lm4857_i2c_driver); platform_device_unregister(neo1973_snd_device); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad381138fc2e..462e635dfc74 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4,8 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * with code, comments and ideas from :- * Richard Purdie <richard@openedhand.com> * @@ -1886,7 +1885,7 @@ module_init(snd_soc_init); module_exit(snd_soc_exit); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk"); MODULE_DESCRIPTION("ALSA SoC Core"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:soc-audio"); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9ca9c08610fa..efbd0b37810a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2,8 +2,7 @@ * soc-dapm.c -- ALSA SoC Dynamic Audio Power Management * * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@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 @@ -1484,6 +1483,26 @@ int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin) EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); /** + * snd_soc_dapm_nc_pin - permanently disable pin. + * @codec: SoC codec + * @pin: pin name + * + * Marks the specified pin as being not connected, disabling it along + * any parent or child widgets. At present this is identical to + * snd_soc_dapm_disable_pin() but in future it will be extended to do + * additional things such as disabling controls which only affect + * paths through the pin. + * + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin) +{ + return snd_soc_dapm_set_pin(codec, pin, 0); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin); + +/** * snd_soc_dapm_get_pin_status - get audio pin status * @codec: audio codec * @pin: audio signal pin endpoint (or start point) @@ -1521,6 +1540,6 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev) EXPORT_SYMBOL_GPL(snd_soc_dapm_free); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk"); MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC"); MODULE_LICENSE("GPL"); |