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June 2010

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[alsa-devel] [PATCH] ASoC: SAMSUNG: Change s3c24xx directory name to samsung
by 최찬우 25 Jun '10

25 Jun '10

This patch changes s3c24xx directory name to samsung. (sound/soc/s3c24xx -> sound/soc/samsung) Currently, the source files related to samsung SoCs(s3c24xx & s3c64xx) are included in sound/soc/s3c24xx. It would be better to change s3c24xx directory name to samsung, because s3c24xx directory will support many board based on samsung SoCs(s5p, etc). Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/Kconfig | 2 +- sound/soc/Makefile | 2 +- sound/soc/s3c24xx/Kconfig | 127 ---- sound/soc/s3c24xx/Makefile | 43 -- sound/soc/s3c24xx/jive_wm8750.c | 196 ------ sound/soc/s3c24xx/lm4857.h | 32 - sound/soc/s3c24xx/ln2440sbc_alc650.c | 85 --- sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 506 --------------- sound/soc/s3c24xx/neo1973_wm8753.c | 707 --------------------- sound/soc/s3c24xx/regs-i2s-v2.h | 115 ---- sound/soc/s3c24xx/s3c-ac97.c | 521 ---------------- sound/soc/s3c24xx/s3c-ac97.h | 23 - sound/soc/s3c24xx/s3c-dma.c | 483 --------------- sound/soc/s3c24xx/s3c-dma.h | 31 - sound/soc/s3c24xx/s3c-i2s-v2.c | 776 ------------------------ sound/soc/s3c24xx/s3c-i2s-v2.h | 103 ---- sound/soc/s3c24xx/s3c-pcm.c | 554 ----------------- sound/soc/s3c24xx/s3c-pcm.h | 123 ---- sound/soc/s3c24xx/s3c2412-i2s.c | 190 ------ sound/soc/s3c24xx/s3c2412-i2s.h | 29 - sound/soc/s3c24xx/s3c24xx-i2s.c | 503 --------------- sound/soc/s3c24xx/s3c24xx-i2s.h | 37 -- sound/soc/s3c24xx/s3c24xx_simtec.c | 394 ------------ sound/soc/s3c24xx/s3c24xx_simtec.h | 22 - sound/soc/s3c24xx/s3c24xx_simtec_hermes.c | 153 ----- sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c | 137 ----- sound/soc/s3c24xx/s3c24xx_uda134x.c | 373 ------------ sound/soc/s3c24xx/s3c64xx-i2s-v4.c | 209 ------- sound/soc/s3c24xx/s3c64xx-i2s.c | 190 ------ sound/soc/s3c24xx/s3c64xx-i2s.h | 42 -- sound/soc/s3c24xx/smdk2443_wm9710.c | 81 --- sound/soc/s3c24xx/smdk64xx_wm8580.c | 266 -------- sound/soc/s3c24xx/smdk_wm9713.c | 94 --- sound/soc/samsung/Kconfig | 127 ++++ sound/soc/samsung/Makefile | 43 ++ sound/soc/samsung/jive_wm8750.c | 196 ++++++ sound/soc/samsung/lm4857.h | 32 + sound/soc/samsung/ln2440sbc_alc650.c | 85 +++ sound/soc/samsung/neo1973_gta02_wm8753.c | 506 +++++++++++++++ sound/soc/samsung/neo1973_wm8753.c | 707 +++++++++++++++++++++ sound/soc/samsung/regs-i2s-v2.h | 115 ++++ sound/soc/samsung/s3c-ac97.c | 521 ++++++++++++++++ sound/soc/samsung/s3c-ac97.h | 23 + sound/soc/samsung/s3c-dma.c | 483 +++++++++++++++ sound/soc/samsung/s3c-dma.h | 31 + sound/soc/samsung/s3c-i2s-v2.c | 776 ++++++++++++++++++++++++ sound/soc/samsung/s3c-i2s-v2.h | 103 ++++ sound/soc/samsung/s3c-pcm.c | 554 +++++++++++++++++ sound/soc/samsung/s3c-pcm.h | 123 ++++ sound/soc/samsung/s3c2412-i2s.c | 190 ++++++ sound/soc/samsung/s3c2412-i2s.h | 29 + sound/soc/samsung/s3c24xx-i2s.c | 503 +++++++++++++++ sound/soc/samsung/s3c24xx-i2s.h | 37 ++ sound/soc/samsung/s3c24xx_simtec.c | 394 ++++++++++++ sound/soc/samsung/s3c24xx_simtec.h | 22 + sound/soc/samsung/s3c24xx_simtec_hermes.c | 153 +++++ sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 137 +++++ sound/soc/samsung/s3c24xx_uda134x.c | 373 ++++++++++++ sound/soc/samsung/s3c64xx-i2s-v4.c | 209 +++++++ sound/soc/samsung/s3c64xx-i2s.c | 190 ++++++ sound/soc/samsung/s3c64xx-i2s.h | 42 ++ sound/soc/samsung/smdk2443_wm9710.c | 81 +++ sound/soc/samsung/smdk64xx_wm8580.c | 266 ++++++++ sound/soc/samsung/smdk_wm9713.c | 94 +++ 64 files changed, 7147 insertions(+), 7147 deletions(-) delete mode 100644 sound/soc/s3c24xx/Kconfig delete mode 100644 sound/soc/s3c24xx/Makefile delete mode 100644 sound/soc/s3c24xx/jive_wm8750.c delete mode 100644 sound/soc/s3c24xx/lm4857.h delete mode 100644 sound/soc/s3c24xx/ln2440sbc_alc650.c delete mode 100644 sound/soc/s3c24xx/neo1973_gta02_wm8753.c delete mode 100644 sound/soc/s3c24xx/neo1973_wm8753.c delete mode 100644 sound/soc/s3c24xx/regs-i2s-v2.h delete mode 100644 sound/soc/s3c24xx/s3c-ac97.c delete mode 100644 sound/soc/s3c24xx/s3c-ac97.h delete mode 100644 sound/soc/s3c24xx/s3c-dma.c delete mode 100644 sound/soc/s3c24xx/s3c-dma.h delete mode 100644 sound/soc/s3c24xx/s3c-i2s-v2.c delete mode 100644 sound/soc/s3c24xx/s3c-i2s-v2.h delete mode 100644 sound/soc/s3c24xx/s3c-pcm.c delete mode 100644 sound/soc/s3c24xx/s3c-pcm.h delete mode 100644 sound/soc/s3c24xx/s3c2412-i2s.c delete mode 100644 sound/soc/s3c24xx/s3c2412-i2s.h delete mode 100644 sound/soc/s3c24xx/s3c24xx-i2s.c delete mode 100644 sound/soc/s3c24xx/s3c24xx-i2s.h delete mode 100644 sound/soc/s3c24xx/s3c24xx_simtec.c delete mode 100644 sound/soc/s3c24xx/s3c24xx_simtec.h delete mode 100644 sound/soc/s3c24xx/s3c24xx_simtec_hermes.c delete mode 100644 sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c delete mode 100644 sound/soc/s3c24xx/s3c24xx_uda134x.c delete mode 100644 sound/soc/s3c24xx/s3c64xx-i2s-v4.c delete mode 100644 sound/soc/s3c24xx/s3c64xx-i2s.c delete mode 100644 sound/soc/s3c24xx/s3c64xx-i2s.h delete mode 100644 sound/soc/s3c24xx/smdk2443_wm9710.c delete mode 100644 sound/soc/s3c24xx/smdk64xx_wm8580.c delete mode 100644 sound/soc/s3c24xx/smdk_wm9713.c create mode 100644 sound/soc/samsung/Kconfig create mode 100644 sound/soc/samsung/Makefile create mode 100644 sound/soc/samsung/jive_wm8750.c create mode 100644 sound/soc/samsung/lm4857.h create mode 100644 sound/soc/samsung/ln2440sbc_alc650.c create mode 100644 sound/soc/samsung/neo1973_gta02_wm8753.c create mode 100644 sound/soc/samsung/neo1973_wm8753.c create mode 100644 sound/soc/samsung/regs-i2s-v2.h create mode 100644 sound/soc/samsung/s3c-ac97.c create mode 100644 sound/soc/samsung/s3c-ac97.h create mode 100644 sound/soc/samsung/s3c-dma.c create mode 100644 sound/soc/samsung/s3c-dma.h create mode 100644 sound/soc/samsung/s3c-i2s-v2.c create mode 100644 sound/soc/samsung/s3c-i2s-v2.h create mode 100644 sound/soc/samsung/s3c-pcm.c create mode 100644 sound/soc/samsung/s3c-pcm.h create mode 100644 sound/soc/samsung/s3c2412-i2s.c create mode 100644 sound/soc/samsung/s3c2412-i2s.h create mode 100644 sound/soc/samsung/s3c24xx-i2s.c create mode 100644 sound/soc/samsung/s3c24xx-i2s.h create mode 100644 sound/soc/samsung/s3c24xx_simtec.c create mode 100644 sound/soc/samsung/s3c24xx_simtec.h create mode 100644 sound/soc/samsung/s3c24xx_simtec_hermes.c create mode 100644 sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c create mode 100644 sound/soc/samsung/s3c24xx_uda134x.c create mode 100644 sound/soc/samsung/s3c64xx-i2s-v4.c create mode 100644 sound/soc/samsung/s3c64xx-i2s.c create mode 100644 sound/soc/samsung/s3c64xx-i2s.h create mode 100644 sound/soc/samsung/smdk2443_wm9710.c create mode 100644 sound/soc/samsung/smdk64xx_wm8580.c create mode 100644 sound/soc/samsung/smdk_wm9713.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index b1749bc..5d467ef 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -32,7 +32,7 @@ source "sound/soc/fsl/Kconfig" source "sound/soc/imx/Kconfig" source "sound/soc/omap/Kconfig" source "sound/soc/pxa/Kconfig" -source "sound/soc/s3c24xx/Kconfig" +source "sound/soc/samsung/Kconfig" source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" source "sound/soc/txx9/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 1470141..9572925 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -10,7 +10,7 @@ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += imx/ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += pxa/ -obj-$(CONFIG_SND_SOC) += s3c24xx/ +obj-$(CONFIG_SND_SOC) += samsung/ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ obj-$(CONFIG_SND_SOC) += txx9/ diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig deleted file mode 100644 index 2a7cc22..0000000 --- a/sound/soc/s3c24xx/Kconfig +++ /dev/null @@ -1,127 +0,0 @@ -config SND_S3C24XX_SOC - tristate "SoC Audio for the Samsung S3CXXXX chips" - depends on ARCH_S3C2410 || ARCH_S3C64XX - select S3C64XX_DMA if ARCH_S3C64XX - help - Say Y or M if you want to add support for codecs attached to - the S3C24XX AC97 or I2S interfaces. You will also need to - select the audio interfaces to support below. - -config SND_S3C24XX_SOC_I2S - tristate - select S3C2410_DMA - -config SND_S3C_I2SV2_SOC - tristate - -config SND_S3C2412_SOC_I2S - tristate - select SND_S3C_I2SV2_SOC - select S3C2410_DMA - -config SND_S3C64XX_SOC_I2S - tristate - select SND_S3C_I2SV2_SOC - select S3C64XX_DMA - -config SND_S3C64XX_SOC_I2S_V4 - tristate - select SND_S3C_I2SV2_SOC - select S3C64XX_DMA - -config SND_S3C_SOC_PCM - tristate - -config SND_S3C_SOC_AC97 - tristate - select SND_SOC_AC97_BUS - -config SND_S3C24XX_SOC_NEO1973_WM8753 - tristate "SoC I2S Audio support for NEO1973 - WM8753" - depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01 - select SND_S3C24XX_SOC_I2S - select SND_SOC_WM8753 - help - Say Y if you want to add support for SoC audio on smdk2440 - with the WM8753. - -config SND_S3C24XX_SOC_NEO1973_GTA02_WM8753 - tristate "Audio support for the Openmoko Neo FreeRunner (GTA02)" - depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA02 - select SND_S3C24XX_SOC_I2S - select SND_SOC_WM8753 - help - This driver provides audio support for the Openmoko Neo FreeRunner - smartphone. - -config SND_S3C24XX_SOC_JIVE_WM8750 - tristate "SoC I2S Audio support for Jive" - depends on SND_S3C24XX_SOC && MACH_JIVE - select SND_SOC_WM8750 - select SND_S3C2412_SOC_I2S - help - Sat Y if you want to add support for SoC audio on the Jive. - -config SND_S3C64XX_SOC_WM8580 - tristate "SoC I2S Audio support for WM8580 on SMDK64XX" - depends on SND_S3C24XX_SOC && MACH_SMDK6410 - select SND_SOC_WM8580 - select SND_S3C64XX_SOC_I2S_V4 - help - Say Y if you want to add support for SoC audio on the SMDK6410. - -config SND_S3C24XX_SOC_SMDK2443_WM9710 - tristate "SoC AC97 Audio support for SMDK2443 - WM9710" - depends on SND_S3C24XX_SOC && MACH_SMDK2443 - select S3C2410_DMA - select AC97_BUS - select SND_SOC_AC97_CODEC - select SND_S3C_SOC_AC97 - help - Say Y if you want to add support for SoC audio on smdk2443 - with the WM9710. - -config SND_S3C24XX_SOC_LN2440SBC_ALC650 - tristate "SoC AC97 Audio support for LN2440SBC - ALC650" - depends on SND_S3C24XX_SOC && ARCH_S3C2410 - select S3C2410_DMA - select AC97_BUS - select SND_SOC_AC97_CODEC - select SND_S3C_SOC_AC97 - help - Say Y if you want to add support for SoC audio on ln2440sbc - with the ALC650. - -config SND_S3C24XX_SOC_S3C24XX_UDA134X - tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" - depends on SND_S3C24XX_SOC && ARCH_S3C2410 - select SND_S3C24XX_SOC_I2S - select SND_SOC_L3 - select SND_SOC_UDA134X - -config SND_S3C24XX_SOC_SIMTEC - tristate - help - Internal node for common S3C24XX/Simtec suppor - -config SND_S3C24XX_SOC_SIMTEC_TLV320AIC23 - tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards" - depends on SND_S3C24XX_SOC && ARCH_S3C2410 - select SND_S3C24XX_SOC_I2S - select SND_SOC_TLV320AIC23 - select SND_S3C24XX_SOC_SIMTEC - -config SND_S3C24XX_SOC_SIMTEC_HERMES - tristate "SoC I2S Audio support for Simtec Hermes board" - depends on SND_S3C24XX_SOC && ARCH_S3C2410 - select SND_S3C24XX_SOC_I2S - select SND_SOC_TLV320AIC3X - select SND_S3C24XX_SOC_SIMTEC - -config SND_SOC_SMDK_WM9713 - tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_S3C24XX_SOC && MACH_SMDK6410 - select SND_SOC_WM9713 - select SND_S3C_SOC_AC97 - help - Sat Y if you want to add support for SoC audio on the SMDK. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile deleted file mode 100644 index 81d8dc5..0000000 --- a/sound/soc/s3c24xx/Makefile +++ /dev/null @@ -1,43 +0,0 @@ -# S3c24XX Platform Support -snd-soc-s3c24xx-objs := s3c-dma.o -snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o -snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o -snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o -snd-soc-s3c-ac97-objs := s3c-ac97.o -snd-soc-s3c64xx-i2s-v4-objs := s3c64xx-i2s-v4.o -snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o -snd-soc-s3c-pcm-objs := s3c-pcm.o - -obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o -obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o -obj-$(CONFIG_SND_S3C_SOC_AC97) += snd-soc-s3c-ac97.o -obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o -obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o -obj-$(CONFIG_SND_S3C64XX_SOC_I2S_V4) += snd-soc-s3c64xx-i2s-v4.o -obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o -obj-$(CONFIG_SND_S3C_SOC_PCM) += snd-soc-s3c-pcm.o - -# S3C24XX Machine Support -snd-soc-jive-wm8750-objs := jive_wm8750.o -snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o -snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o -snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o -snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o -snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o -snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o -snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o -snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o -snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o -snd-soc-smdk-wm9713-objs := smdk_wm9713.o - -obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o -obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o -obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o -obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o -obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o -obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o -obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o -obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o -obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o -obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o -obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c deleted file mode 100644 index 8c108b1..0000000 --- a/sound/soc/s3c24xx/jive_wm8750.c +++ /dev/null @@ -1,196 +0,0 @@ -/* sound/soc/s3c24xx/jive_wm8750.c - * - * Copyright 2007,2008 Simtec Electronics - * - * Based on sound/soc/pxa/spitz.c - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand 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. -*/ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/clk.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <asm/mach-types.h> - -#include "s3c-dma.h" -#include "s3c2412-i2s.h" - -#include "../codecs/wm8750.h" - -static const struct snd_soc_dapm_route audio_map[] = { - { "Headphone Jack", NULL, "LOUT1" }, - { "Headphone Jack", NULL, "ROUT1" }, - { "Internal Speaker", NULL, "LOUT2" }, - { "Internal Speaker", NULL, "ROUT2" }, - { "LINPUT1", NULL, "Line Input" }, - { "RINPUT1", NULL, "Line Input" }, -}; - -static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Internal Speaker", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), -}; - -static int jive_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; - struct s3c_i2sv2_rate_calc div; - unsigned int clk = 0; - int ret = 0; - - switch (params_rate(params)) { - case 8000: - case 16000: - case 48000: - case 96000: - clk = 12288000; - break; - case 11025: - case 22050: - case 44100: - clk = 11289600; - break; - } - - s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), - s3c_i2sv2_get_clock(cpu_dai)); - - /* 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; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_RCLK, div.fs_div); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_PRESCALER, - div.clk_div - 1); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops jive_ops = { - .hw_params = jive_hw_params, -}; - -static int jive_wm8750_init(struct snd_soc_codec *codec) -{ - int err; - - /* These endpoints are not being used. */ - 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, "MONO"); - - /* Add jive specific widgets */ - err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, - ARRAY_SIZE(wm8750_dapm_widgets)); - if (err) { - printk(KERN_ERR "%s: failed to add widgets (%d)\n", - __func__, err); - return err; - } - - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(codec); - - return 0; -} - -static struct snd_soc_dai_link jive_dai = { - .name = "wm8750", - .stream_name = "WM8750", - .cpu_dai = &s3c2412_i2s_dai, - .codec_dai = &wm8750_dai, - .init = jive_wm8750_init, - .ops = &jive_ops, -}; - -/* jive audio machine driver */ -static struct snd_soc_card snd_soc_machine_jive = { - .name = "Jive", - .platform = &s3c24xx_soc_platform, - .dai_link = &jive_dai, - .num_links = 1, -}; - -/* jive audio subsystem */ -static struct snd_soc_device jive_snd_devdata = { - .card = &snd_soc_machine_jive, - .codec_dev = &soc_codec_dev_wm8750, -}; - -static struct platform_device *jive_snd_device; - -static int __init jive_init(void) -{ - int ret; - - if (!machine_is_jive()) - return 0; - - printk("JIVE WM8750 Audio support\n"); - - jive_snd_device = platform_device_alloc("soc-audio", -1); - if (!jive_snd_device) - return -ENOMEM; - - platform_set_drvdata(jive_snd_device, &jive_snd_devdata); - jive_snd_devdata.dev = &jive_snd_device->dev; - ret = platform_device_add(jive_snd_device); - - if (ret) - platform_device_put(jive_snd_device); - - return ret; -} - -static void __exit jive_exit(void) -{ - platform_device_unregister(jive_snd_device); -} - -module_init(jive_init); -module_exit(jive_exit); - -MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("ALSA SoC Jive Audio support"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/lm4857.h b/sound/soc/s3c24xx/lm4857.h deleted file mode 100644 index 0cf5b70..0000000 --- a/sound/soc/s3c24xx/lm4857.h +++ /dev/null @@ -1,32 +0,0 @@ -/* - * lm4857.h -- ALSA Soc Audio Layer - * - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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 - * 18th Jun 2007 Initial version. - */ - -#ifndef LM4857_H_ -#define LM4857_H_ - -/* The register offsets in the cache array */ -#define LM4857_MVOL 0 -#define LM4857_LVOL 1 -#define LM4857_RVOL 2 -#define LM4857_CTRL 3 - -/* the shifts required to set these bits */ -#define LM4857_3D 5 -#define LM4857_WAKEUP 5 -#define LM4857_EPGAIN 4 - -#endif /*LM4857_H_*/ - diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c deleted file mode 100644 index ffa954f..0000000 --- a/sound/soc/s3c24xx/ln2440sbc_alc650.c +++ /dev/null @@ -1,85 +0,0 @@ -/* - * SoC audio for ln2440sbc - * - * Copyright 2007 KonekTel, a.s. - * Author: Ivan Kuten - * ivan.kuten(a)promwad.com - * - * Heavily based on smdk2443_wm9710.c - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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. - * - */ - -#include <linux/module.h> -#include <linux/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include "../codecs/ac97.h" -#include "s3c-dma.h" -#include "s3c-ac97.h" - -static struct snd_soc_card ln2440sbc; - -static struct snd_soc_dai_link ln2440sbc_dai[] = { -{ - .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], - .codec_dai = &ac97_dai, -}, -}; - -static struct snd_soc_card ln2440sbc = { - .name = "LN2440SBC", - .platform = &s3c24xx_soc_platform, - .dai_link = ln2440sbc_dai, - .num_links = ARRAY_SIZE(ln2440sbc_dai), -}; - -static struct snd_soc_device ln2440sbc_snd_ac97_devdata = { - .card = &ln2440sbc, - .codec_dev = &soc_codec_dev_ac97, -}; - -static struct platform_device *ln2440sbc_snd_ac97_device; - -static int __init ln2440sbc_init(void) -{ - int ret; - - ln2440sbc_snd_ac97_device = platform_device_alloc("soc-audio", -1); - if (!ln2440sbc_snd_ac97_device) - return -ENOMEM; - - platform_set_drvdata(ln2440sbc_snd_ac97_device, - &ln2440sbc_snd_ac97_devdata); - ln2440sbc_snd_ac97_devdata.dev = &ln2440sbc_snd_ac97_device->dev; - ret = platform_device_add(ln2440sbc_snd_ac97_device); - - if (ret) - platform_device_put(ln2440sbc_snd_ac97_device); - - return ret; -} - -static void __exit ln2440sbc_exit(void) -{ - platform_device_unregister(ln2440sbc_snd_ac97_device); -} - -module_init(ln2440sbc_init); -module_exit(ln2440sbc_exit); - -/* Module information */ -MODULE_AUTHOR("Ivan Kuten"); -MODULE_DESCRIPTION("ALSA SoC ALC650 LN2440SBC"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c deleted file mode 100644 index 209c259..0000000 --- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c +++ /dev/null @@ -1,506 +0,0 @@ -/* - * neo1973_gta02_wm8753.c -- SoC audio for Openmoko Freerunner(GTA02) - * - * Copyright 2007 Openmoko Inc - * Author: Graeme Gregory <graeme(a)openmoko.org> - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory <linux(a)wolfsonmicro.com> - * Copyright 2009 Wolfson Microelectronics - * - * 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/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/gpio.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <asm/mach-types.h> - -#include <plat/regs-iis.h> - -#include <mach/regs-clock.h> -#include <asm/io.h> -#include <mach/gta02.h> -#include "../codecs/wm8753.h" -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" - -static struct snd_soc_card neo1973_gta02; - -static int neo1973_gta02_hifi_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; - unsigned int pll_out = 0, bclk = 0; - int ret = 0; - unsigned long iis_clkrate; - - iis_clkrate = s3c24xx_i2s_get_clockrate(); - - switch (params_rate(params)) { - case 8000: - case 16000: - pll_out = 12288000; - break; - case 48000: - bclk = WM8753_BCLK_DIV_4; - pll_out = 12288000; - break; - case 96000: - bclk = WM8753_BCLK_DIV_2; - pll_out = 12288000; - break; - case 11025: - bclk = WM8753_BCLK_DIV_16; - pll_out = 11289600; - break; - case 22050: - bclk = WM8753_BCLK_DIV_8; - pll_out = 11289600; - break; - case 44100: - bclk = WM8753_BCLK_DIV_4; - pll_out = 11289600; - break; - case 88200: - bclk = WM8753_BCLK_DIV_2; - pll_out = 11289600; - break; - } - - /* 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; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set MCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, - S3C2410_IISMOD_32FS); - if (ret < 0) - return ret; - - /* set codec BCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(codec_dai, - WM8753_BCLKDIV, bclk); - if (ret < 0) - return ret; - - /* set prescaler division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - S3C24XX_PRESCALE(4, 4)); - if (ret < 0) - return ret; - - /* codec PLL input is PCLK/4 */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, - iis_clkrate / 4, pll_out); - if (ret < 0) - return ret; - - return 0; -} - -static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - - /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); -} - -/* - * Neo1973 WM8753 HiFi DAI opserations. - */ -static struct snd_soc_ops neo1973_gta02_hifi_ops = { - .hw_params = neo1973_gta02_hifi_hw_params, - .hw_free = neo1973_gta02_hifi_hw_free, -}; - -static int neo1973_gta02_voice_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; - unsigned int pcmdiv = 0; - int ret = 0; - unsigned long iis_clkrate; - - iis_clkrate = s3c24xx_i2s_get_clockrate(); - - if (params_rate(params) != 8000) - return -EINVAL; - if (params_channels(params) != 1) - return -EINVAL; - - pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ - - /* todo: gg check mode (DSP_B) against CSR datasheet */ - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, - 12288000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set codec PCM division for sample rate */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, - pcmdiv); - if (ret < 0) - return ret; - - /* configue and enable PLL for 12.288MHz output */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, - iis_clkrate / 4, 12288000); - if (ret < 0) - return ret; - - return 0; -} - -static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - - /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); -} - -static struct snd_soc_ops neo1973_gta02_voice_ops = { - .hw_params = neo1973_gta02_voice_hw_params, - .hw_free = neo1973_gta02_voice_hw_free, -}; - -#define LM4853_AMP 1 -#define LM4853_SPK 2 - -static u8 lm4853_state; - -/* This has no effect, it exists only to maintain compatibility with - * existing ALSA state files. - */ -static int lm4853_set_state(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int val = ucontrol->value.integer.value[0]; - - if (val) - lm4853_state |= LM4853_AMP; - else - lm4853_state &= ~LM4853_AMP; - - return 0; -} - -static int lm4853_get_state(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = lm4853_state & LM4853_AMP; - - return 0; -} - -static int lm4853_set_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int val = ucontrol->value.integer.value[0]; - - if (val) { - lm4853_state |= LM4853_SPK; - gpio_set_value(GTA02_GPIO_HP_IN, 0); - } else { - lm4853_state &= ~LM4853_SPK; - gpio_set_value(GTA02_GPIO_HP_IN, 1); - } - - return 0; -} - -static int lm4853_get_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = (lm4853_state & LM4853_SPK) >> 1; - - return 0; -} - -static int lm4853_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, - int event) -{ - gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(value)); - - return 0; -} - -static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Stereo Out", lm4853_event), - SND_SOC_DAPM_LINE("GSM Line Out", NULL), - SND_SOC_DAPM_LINE("GSM Line In", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Handset Mic", NULL), - SND_SOC_DAPM_SPK("Handset Spk", NULL), -}; - - -/* example machine audio_mapnections */ -static const struct snd_soc_dapm_route audio_map[] = { - - /* Connections to the lm4853 amp */ - {"Stereo Out", NULL, "LOUT1"}, - {"Stereo Out", NULL, "ROUT1"}, - - /* Connections to the GSM Module */ - {"GSM Line Out", NULL, "MONO1"}, - {"GSM Line Out", NULL, "MONO2"}, - {"RXP", NULL, "GSM Line In"}, - {"RXN", NULL, "GSM Line In"}, - - /* Connections to Headset */ - {"MIC1", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Headset Mic"}, - - /* Call Mic */ - {"MIC2", NULL, "Mic Bias"}, - {"MIC2N", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Handset Mic"}, - - /* Call Speaker */ - {"Handset Spk", NULL, "LOUT2"}, - {"Handset Spk", NULL, "ROUT2"}, - - /* Connect the ALC pins */ - {"ACIN", NULL, "ACOP"}, -}; - -static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = { - SOC_DAPM_PIN_SWITCH("Stereo Out"), - SOC_DAPM_PIN_SWITCH("GSM Line Out"), - SOC_DAPM_PIN_SWITCH("GSM Line In"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Handset Mic"), - SOC_DAPM_PIN_SWITCH("Handset Spk"), - - /* This has no effect, it exists only to maintain compatibility with - * existing ALSA state files. - */ - SOC_SINGLE_EXT("Amp State Switch", 6, 0, 1, 0, - lm4853_get_state, - lm4853_set_state), - SOC_SINGLE_EXT("Amp Spk Switch", 7, 0, 1, 0, - lm4853_get_spk, - lm4853_set_spk), -}; - -/* - * This is an example machine initialisation for a wm8753 connected to a - * neo1973 GTA02. - */ -static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec) -{ - int err; - - /* set up NC codec pins */ - 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 gta02 specific widgets */ - snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, - ARRAY_SIZE(wm8753_dapm_widgets)); - - /* add neo1973 gta02 specific controls */ - err = snd_soc_add_controls(codec, wm8753_neo1973_gta02_controls, - ARRAY_SIZE(wm8753_neo1973_gta02_controls)); - - if (err < 0) - return err; - - /* set up neo1973 gta02 specific audio path audio_map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - - /* set endpoints to default off mode */ - snd_soc_dapm_disable_pin(codec, "Stereo Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Handset Mic"); - snd_soc_dapm_disable_pin(codec, "Handset Spk"); - - /* allow audio paths from the GSM modem to run during suspend */ - snd_soc_dapm_ignore_suspend(codec, "Stereo Out"); - snd_soc_dapm_ignore_suspend(codec, "GSM Line Out"); - snd_soc_dapm_ignore_suspend(codec, "GSM Line In"); - snd_soc_dapm_ignore_suspend(codec, "Headset Mic"); - snd_soc_dapm_ignore_suspend(codec, "Handset Mic"); - snd_soc_dapm_ignore_suspend(codec, "Handset Spk"); - - snd_soc_dapm_sync(codec); - - return 0; -} - -/* - * BT Codec DAI - */ -static struct snd_soc_dai bt_dai = { - .name = "Bluetooth", - .id = 0, - .playback = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, -}; - -static struct snd_soc_dai_link neo1973_gta02_dai[] = { -{ /* Hifi Playback - for similatious use with voice below */ - .name = "WM8753", - .stream_name = "WM8753 HiFi", - .cpu_dai = &s3c24xx_i2s_dai, - .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], - .init = neo1973_gta02_wm8753_init, - .ops = &neo1973_gta02_hifi_ops, -}, -{ /* Voice via BT */ - .name = "Bluetooth", - .stream_name = "Voice", - .cpu_dai = &bt_dai, - .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], - .ops = &neo1973_gta02_voice_ops, -}, -}; - -static struct snd_soc_card neo1973_gta02 = { - .name = "neo1973-gta02", - .platform = &s3c24xx_soc_platform, - .dai_link = neo1973_gta02_dai, - .num_links = ARRAY_SIZE(neo1973_gta02_dai), -}; - -static struct snd_soc_device neo1973_gta02_snd_devdata = { - .card = &neo1973_gta02, - .codec_dev = &soc_codec_dev_wm8753, -}; - -static struct platform_device *neo1973_gta02_snd_device; - -static int __init neo1973_gta02_init(void) -{ - int ret; - - if (!machine_is_neo1973_gta02()) { - printk(KERN_INFO - "Only GTA02 is supported by this ASoC driver\n"); - return -ENODEV; - } - - /* register bluetooth DAI here */ - ret = snd_soc_register_dai(&bt_dai); - if (ret) - return ret; - - neo1973_gta02_snd_device = platform_device_alloc("soc-audio", -1); - if (!neo1973_gta02_snd_device) - return -ENOMEM; - - platform_set_drvdata(neo1973_gta02_snd_device, - &neo1973_gta02_snd_devdata); - neo1973_gta02_snd_devdata.dev = &neo1973_gta02_snd_device->dev; - ret = platform_device_add(neo1973_gta02_snd_device); - - if (ret) { - platform_device_put(neo1973_gta02_snd_device); - return ret; - } - - /* Initialise GPIOs used by amp */ - ret = gpio_request(GTA02_GPIO_HP_IN, "GTA02_HP_IN"); - if (ret) { - pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_HP_IN); - goto err_unregister_device; - } - - ret = gpio_direction_output(GTA02_GPIO_AMP_HP_IN, 1); - if (ret) { - pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_HP_IN); - goto err_free_gpio_hp_in; - } - - ret = gpio_request(GTA02_GPIO_AMP_SHUT, "GTA02_AMP_SHUT"); - if (ret) { - pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_AMP_SHUT); - goto err_free_gpio_hp_in; - } - - ret = gpio_direction_output(GTA02_GPIO_AMP_SHUT, 1); - if (ret) { - pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_AMP_SHUT); - goto err_free_gpio_amp_shut; - } - - return 0; - -err_free_gpio_amp_shut: - gpio_free(GTA02_GPIO_AMP_SHUT); -err_free_gpio_hp_in: - gpio_free(GTA02_GPIO_HP_IN); -err_unregister_device: - platform_device_unregister(neo1973_gta02_snd_device); - return ret; -} -module_init(neo1973_gta02_init); - -static void __exit neo1973_gta02_exit(void) -{ - snd_soc_unregister_dai(&bt_dai); - platform_device_unregister(neo1973_gta02_snd_device); - gpio_free(GTA02_GPIO_HP_IN); - gpio_free(GTA02_GPIO_AMP_SHUT); -} -module_exit(neo1973_gta02_exit); - -/* Module information */ -MODULE_AUTHOR("Graeme Gregory, graeme(a)openmoko.org"); -MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 GTA02"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c deleted file mode 100644 index 0cb4f86..0000000 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ /dev/null @@ -1,707 +0,0 @@ -/* - * neo1973_wm8753.c -- SoC audio for Neo1973 - * - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/i2c.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> -#include <sound/tlv.h> - -#include <asm/mach-types.h> -#include <asm/hardware/scoop.h> -#include <mach/regs-clock.h> -#include <mach/regs-gpio.h> -#include <mach/hardware.h> -#include <linux/io.h> -#include <mach/spi-gpio.h> - -#include <plat/regs-iis.h> - -#include "../codecs/wm8753.h" -#include "lm4857.h" -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" - -/* define the scenarios */ -#define NEO_AUDIO_OFF 0 -#define NEO_GSM_CALL_AUDIO_HANDSET 1 -#define NEO_GSM_CALL_AUDIO_HEADSET 2 -#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3 -#define NEO_STEREO_TO_SPEAKERS 4 -#define NEO_STEREO_TO_HEADPHONES 5 -#define NEO_CAPTURE_HANDSET 6 -#define NEO_CAPTURE_HEADSET 7 -#define NEO_CAPTURE_BLUETOOTH 8 - -static struct snd_soc_card neo1973; -static struct i2c_client *i2c; - -static int neo1973_hifi_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; - unsigned int pll_out = 0, bclk = 0; - int ret = 0; - unsigned long iis_clkrate; - - pr_debug("Entered %s\n", __func__); - - iis_clkrate = s3c24xx_i2s_get_clockrate(); - - switch (params_rate(params)) { - case 8000: - case 16000: - pll_out = 12288000; - break; - case 48000: - bclk = WM8753_BCLK_DIV_4; - pll_out = 12288000; - break; - case 96000: - bclk = WM8753_BCLK_DIV_2; - pll_out = 12288000; - break; - case 11025: - bclk = WM8753_BCLK_DIV_16; - pll_out = 11289600; - break; - case 22050: - bclk = WM8753_BCLK_DIV_8; - pll_out = 11289600; - break; - case 44100: - bclk = WM8753_BCLK_DIV_4; - pll_out = 11289600; - break; - case 88200: - bclk = WM8753_BCLK_DIV_2; - pll_out = 11289600; - break; - } - - /* 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; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set MCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, - S3C2410_IISMOD_32FS); - if (ret < 0) - return ret; - - /* set codec BCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); - if (ret < 0) - return ret; - - /* set prescaler division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - S3C24XX_PRESCALE(4, 4)); - if (ret < 0) - return ret; - - /* codec PLL input is PCLK/4 */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, - iis_clkrate / 4, pll_out); - if (ret < 0) - return ret; - - return 0; -} - -static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - - pr_debug("Entered %s\n", __func__); - - /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); -} - -/* - * Neo1973 WM8753 HiFi DAI opserations. - */ -static struct snd_soc_ops neo1973_hifi_ops = { - .hw_params = neo1973_hifi_hw_params, - .hw_free = neo1973_hifi_hw_free, -}; - -static int neo1973_voice_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; - unsigned int pcmdiv = 0; - int ret = 0; - unsigned long iis_clkrate; - - pr_debug("Entered %s\n", __func__); - - iis_clkrate = s3c24xx_i2s_get_clockrate(); - - if (params_rate(params) != 8000) - return -EINVAL; - if (params_channels(params) != 1) - return -EINVAL; - - pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ - - /* todo: gg check mode (DSP_B) against CSR datasheet */ - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set codec PCM division for sample rate */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); - if (ret < 0) - return ret; - - /* configue and enable PLL for 12.288MHz output */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, - iis_clkrate / 4, 12288000); - if (ret < 0) - return ret; - - return 0; -} - -static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - - pr_debug("Entered %s\n", __func__); - - /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); -} - -static struct snd_soc_ops neo1973_voice_ops = { - .hw_params = neo1973_voice_hw_params, - .hw_free = neo1973_voice_hw_free, -}; - -static int neo1973_scenario; - -static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = neo1973_scenario; - return 0; -} - -static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) -{ - pr_debug("Entered %s\n", __func__); - - switch (neo1973_scenario) { - case NEO_AUDIO_OFF: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_HANDSET: - snd_soc_dapm_enable_pin(codec, "Audio Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_enable_pin(codec, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_HEADSET: - snd_soc_dapm_enable_pin(codec, "Audio Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line In"); - snd_soc_dapm_enable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_BLUETOOTH: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - case NEO_STEREO_TO_SPEAKERS: - snd_soc_dapm_enable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - case NEO_STEREO_TO_HEADPHONES: - snd_soc_dapm_enable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - case NEO_CAPTURE_HANDSET: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_enable_pin(codec, "Call Mic"); - break; - case NEO_CAPTURE_HEADSET: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_enable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - case NEO_CAPTURE_BLUETOOTH: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - default: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - } - - snd_soc_dapm_sync(codec); - - return 0; -} - -static int neo1973_set_scenario(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - - pr_debug("Entered %s\n", __func__); - - if (neo1973_scenario == ucontrol->value.integer.value[0]) - return 0; - - neo1973_scenario = ucontrol->value.integer.value[0]; - set_scenario_endpoints(codec, neo1973_scenario); - return 1; -} - -static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; - -static void lm4857_write_regs(void) -{ - pr_debug("Entered %s\n", __func__); - - if (i2c_master_send(i2c, lm4857_regs, 4) != 4) - printk(KERN_ERR "lm4857: i2c write failed\n"); -} - -static int lm4857_get_reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int reg = mc->reg; - int shift = mc->shift; - int mask = mc->max; - - pr_debug("Entered %s\n", __func__); - - ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask; - return 0; -} - -static int lm4857_set_reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int reg = mc->reg; - int shift = mc->shift; - int mask = mc->max; - - if (((lm4857_regs[reg] >> shift) & mask) == - ucontrol->value.integer.value[0]) - return 0; - - lm4857_regs[reg] &= ~(mask << shift); - lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift; - lm4857_write_regs(); - return 1; -} - -static int lm4857_get_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - u8 value = lm4857_regs[LM4857_CTRL] & 0x0F; - - pr_debug("Entered %s\n", __func__); - - if (value) - value -= 5; - - ucontrol->value.integer.value[0] = value; - return 0; -} - -static int lm4857_set_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - u8 value = ucontrol->value.integer.value[0]; - - pr_debug("Entered %s\n", __func__); - - if (value) - value += 5; - - if ((lm4857_regs[LM4857_CTRL] & 0x0F) == value) - return 0; - - lm4857_regs[LM4857_CTRL] &= 0xF0; - lm4857_regs[LM4857_CTRL] |= value; - lm4857_write_regs(); - return 1; -} - -static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { - SND_SOC_DAPM_LINE("Audio Out", NULL), - SND_SOC_DAPM_LINE("GSM Line Out", NULL), - SND_SOC_DAPM_LINE("GSM Line In", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Call Mic", NULL), -}; - - -static const struct snd_soc_dapm_route dapm_routes[] = { - - /* Connections to the lm4857 amp */ - {"Audio Out", NULL, "LOUT1"}, - {"Audio Out", NULL, "ROUT1"}, - - /* Connections to the GSM Module */ - {"GSM Line Out", NULL, "MONO1"}, - {"GSM Line Out", NULL, "MONO2"}, - {"RXP", NULL, "GSM Line In"}, - {"RXN", NULL, "GSM Line In"}, - - /* Connections to Headset */ - {"MIC1", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Headset Mic"}, - - /* Call Mic */ - {"MIC2", NULL, "Mic Bias"}, - {"MIC2N", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Call Mic"}, - - /* Connect the ALC pins */ - {"ACIN", NULL, "ACOP"}, -}; - -static const char *lm4857_mode[] = { - "Off", - "Call Speaker", - "Stereo Speakers", - "Stereo Speakers + Headphones", - "Headphones" -}; - -static const struct soc_enum lm4857_mode_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode), -}; - -static const char *neo_scenarios[] = { - "Off", - "GSM Handset", - "GSM Headset", - "GSM Bluetooth", - "Speakers", - "Headphones", - "Capture Handset", - "Capture Headset", - "Capture Bluetooth" -}; - -static const struct soc_enum neo_scenario_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios), -}; - -static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); -static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); - -static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { - SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg, stereo_tlv), - SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg, stereo_tlv), - SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg, mono_tlv), - SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], - lm4857_get_mode, lm4857_set_mode), - SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0], - neo1973_get_scenario, neo1973_set_scenario), - SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Fast Wakeup Playback Switch", LM4857_CTRL, 5, 1, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Earpiece 6dB Playback Switch", LM4857_CTRL, 4, 1, 0, - lm4857_get_reg, lm4857_set_reg), -}; - -/* - * This is an example machine initialisation for a wm8753 connected to a - * neo1973 II. It is missing logic to detect hp/mic insertions and logic - * to re-route the audio in such an event. - */ -static int neo1973_wm8753_init(struct snd_soc_codec *codec) -{ - int err; - - pr_debug("Entered %s\n", __func__); - - /* set up NC codec pins */ - 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 */ - err = snd_soc_add_controls(codec, wm8753_neo1973_controls, - ARRAY_SIZE(8753_neo1973_controls)); - if (err < 0) - return err; - - /* set up neo1973 specific audio routes */ - err = snd_soc_dapm_add_routes(codec, dapm_routes, - ARRAY_SIZE(dapm_routes)); - - snd_soc_dapm_sync(codec); - return 0; -} - -/* - * BT Codec DAI - */ -static struct snd_soc_dai bt_dai = { - .name = "Bluetooth", - .id = 0, - .playback = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, -}; - -static struct snd_soc_dai_link neo1973_dai[] = { -{ /* Hifi Playback - for similatious use with voice below */ - .name = "WM8753", - .stream_name = "WM8753 HiFi", - .cpu_dai = &s3c24xx_i2s_dai, - .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], - .init = neo1973_wm8753_init, - .ops = &neo1973_hifi_ops, -}, -{ /* Voice via BT */ - .name = "Bluetooth", - .stream_name = "Voice", - .cpu_dai = &bt_dai, - .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], - .ops = &neo1973_voice_ops, -}, -}; - -static struct snd_soc_card neo1973 = { - .name = "neo1973", - .platform = &s3c24xx_soc_platform, - .dai_link = neo1973_dai, - .num_links = ARRAY_SIZE(neo1973_dai), -}; - -static struct snd_soc_device neo1973_snd_devdata = { - .card = &neo1973, - .codec_dev = &soc_codec_dev_wm8753, -}; - -static int lm4857_i2c_probe(struct i2c_client *client, - const struct i2c_device_id *id) -{ - pr_debug("Entered %s\n", __func__); - - i2c = client; - - lm4857_write_regs(); - return 0; -} - -static int lm4857_i2c_remove(struct i2c_client *client) -{ - pr_debug("Entered %s\n", __func__); - - i2c = NULL; - - return 0; -} - -static u8 lm4857_state; - -static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) -{ - pr_debug("Entered %s\n", __func__); - - dev_dbg(&dev->dev, "lm4857_suspend\n"); - lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf; - if (lm4857_state) { - lm4857_regs[LM4857_CTRL] &= 0xf0; - lm4857_write_regs(); - } - return 0; -} - -static int lm4857_resume(struct i2c_client *dev) -{ - pr_debug("Entered %s\n", __func__); - - if (lm4857_state) { - lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f); - lm4857_write_regs(); - } - return 0; -} - -static void lm4857_shutdown(struct i2c_client *dev) -{ - pr_debug("Entered %s\n", __func__); - - dev_dbg(&dev->dev, "lm4857_shutdown\n"); - lm4857_regs[LM4857_CTRL] &= 0xf0; - lm4857_write_regs(); -} - -static const struct i2c_device_id lm4857_i2c_id[] = { - { "neo1973_lm4857", 0 }, - { } -}; - -static struct i2c_driver lm4857_i2c_driver = { - .driver = { - .name = "LM4857 I2C Amp", - .owner = THIS_MODULE, - }, - .suspend = lm4857_suspend, - .resume = lm4857_resume, - .shutdown = lm4857_shutdown, - .probe = lm4857_i2c_probe, - .remove = lm4857_i2c_remove, - .id_table = lm4857_i2c_id, -}; - -static struct platform_device *neo1973_snd_device; - -static int __init neo1973_init(void) -{ - int ret; - - pr_debug("Entered %s\n", __func__); - - if (!machine_is_neo1973_gta01()) { - printk(KERN_INFO - "Only GTA01 hardware supported by ASoC driver\n"); - return -ENODEV; - } - - neo1973_snd_device = platform_device_alloc("soc-audio", -1); - if (!neo1973_snd_device) - return -ENOMEM; - - platform_set_drvdata(neo1973_snd_device, &neo1973_snd_devdata); - neo1973_snd_devdata.dev = &neo1973_snd_device->dev; - ret = platform_device_add(neo1973_snd_device); - - if (ret) { - platform_device_put(neo1973_snd_device); - return ret; - } - - ret = i2c_add_driver(&lm4857_i2c_driver); - - if (ret != 0) - platform_device_unregister(neo1973_snd_device); - - return ret; -} - -static void __exit neo1973_exit(void) -{ - pr_debug("Entered %s\n", __func__); - - i2c_del_driver(&lm4857_i2c_driver); - platform_device_unregister(neo1973_snd_device); -} - -module_init(neo1973_init); -module_exit(neo1973_exit); - -/* Module information */ -MODULE_AUTHOR("Graeme Gregory, graeme(a)openmoko.org, www.openmoko.org"); -MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/regs-i2s-v2.h b/sound/soc/s3c24xx/regs-i2s-v2.h deleted file mode 100644 index 5e5e568..0000000 --- a/sound/soc/s3c24xx/regs-i2s-v2.h +++ /dev/null @@ -1,115 +0,0 @@ -/* linux/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h - * - * Copyright 2007 Simtec Electronics <linux(a)simtec.co.uk> - * http://armlinux.simtec.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 - * published by the Free Software Foundation. - * - * S3C2412 IIS register definition -*/ - -#ifndef __ASM_ARCH_REGS_S3C2412_IIS_H -#define __ASM_ARCH_REGS_S3C2412_IIS_H - -#define S3C2412_IISCON (0x00) -#define S3C2412_IISMOD (0x04) -#define S3C2412_IISFIC (0x08) -#define S3C2412_IISPSR (0x0C) -#define S3C2412_IISTXD (0x10) -#define S3C2412_IISRXD (0x14) - -#define S5PC1XX_IISFICS 0x18 -#define S5PC1XX_IISTXDS 0x1C - -#define S5PC1XX_IISCON_SW_RST (1 << 31) -#define S5PC1XX_IISCON_FRXOFSTATUS (1 << 26) -#define S5PC1XX_IISCON_FRXORINTEN (1 << 25) -#define S5PC1XX_IISCON_FTXSURSTAT (1 << 24) -#define S5PC1XX_IISCON_FTXSURINTEN (1 << 23) -#define S5PC1XX_IISCON_TXSDMAPAUSE (1 << 20) -#define S5PC1XX_IISCON_TXSDMACTIVE (1 << 18) - -#define S3C64XX_IISCON_FTXURSTATUS (1 << 17) -#define S3C64XX_IISCON_FTXURINTEN (1 << 16) -#define S3C64XX_IISCON_TXFIFO2_EMPTY (1 << 15) -#define S3C64XX_IISCON_TXFIFO1_EMPTY (1 << 14) -#define S3C64XX_IISCON_TXFIFO2_FULL (1 << 13) -#define S3C64XX_IISCON_TXFIFO1_FULL (1 << 12) - -#define S3C2412_IISCON_LRINDEX (1 << 11) -#define S3C2412_IISCON_TXFIFO_EMPTY (1 << 10) -#define S3C2412_IISCON_RXFIFO_EMPTY (1 << 9) -#define S3C2412_IISCON_TXFIFO_FULL (1 << 8) -#define S3C2412_IISCON_RXFIFO_FULL (1 << 7) -#define S3C2412_IISCON_TXDMA_PAUSE (1 << 6) -#define S3C2412_IISCON_RXDMA_PAUSE (1 << 5) -#define S3C2412_IISCON_TXCH_PAUSE (1 << 4) -#define S3C2412_IISCON_RXCH_PAUSE (1 << 3) -#define S3C2412_IISCON_TXDMA_ACTIVE (1 << 2) -#define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) -#define S3C2412_IISCON_IIS_ACTIVE (1 << 0) - -#define S5PC1XX_IISMOD_OPCLK_CDCLK_OUT (0 << 30) -#define S5PC1XX_IISMOD_OPCLK_CDCLK_IN (1 << 30) -#define S5PC1XX_IISMOD_OPCLK_BCLK_OUT (2 << 30) -#define S5PC1XX_IISMOD_OPCLK_PCLK (3 << 30) -#define S5PC1XX_IISMOD_OPCLK_MASK (3 << 30) -#define S5PC1XX_IISMOD_TXS_IDMA (1 << 28) /* Sec_TXFIFO use I-DMA */ -#define S5PC1XX_IISMOD_BLCS_MASK 0x3 -#define S5PC1XX_IISMOD_BLCS_SHIFT 26 -#define S5PC1XX_IISMOD_BLCP_MASK 0x3 -#define S5PC1XX_IISMOD_BLCP_SHIFT 24 - -#define S3C64XX_IISMOD_C2DD_HHALF (1 << 21) /* Discard Higher-half */ -#define S3C64XX_IISMOD_C2DD_LHALF (1 << 20) /* Discard Lower-half */ -#define S3C64XX_IISMOD_C1DD_HHALF (1 << 19) -#define S3C64XX_IISMOD_C1DD_LHALF (1 << 18) -#define S3C64XX_IISMOD_DC2_EN (1 << 17) -#define S3C64XX_IISMOD_DC1_EN (1 << 16) -#define S3C64XX_IISMOD_BLC_16BIT (0 << 13) -#define S3C64XX_IISMOD_BLC_8BIT (1 << 13) -#define S3C64XX_IISMOD_BLC_24BIT (2 << 13) -#define S3C64XX_IISMOD_BLC_MASK (3 << 13) - -#define S3C2412_IISMOD_IMS_SYSMUX (1 << 10) -#define S3C2412_IISMOD_SLAVE (1 << 11) -#define S3C2412_IISMOD_MODE_TXONLY (0 << 8) -#define S3C2412_IISMOD_MODE_RXONLY (1 << 8) -#define S3C2412_IISMOD_MODE_TXRX (2 << 8) -#define S3C2412_IISMOD_MODE_MASK (3 << 8) -#define S3C2412_IISMOD_LR_LLOW (0 << 7) -#define S3C2412_IISMOD_LR_RLOW (1 << 7) -#define S3C2412_IISMOD_SDF_IIS (0 << 5) -#define S3C2412_IISMOD_SDF_MSB (1 << 5) -#define S3C2412_IISMOD_SDF_LSB (2 << 5) -#define S3C2412_IISMOD_SDF_MASK (3 << 5) -#define S3C2412_IISMOD_RCLK_256FS (0 << 3) -#define S3C2412_IISMOD_RCLK_512FS (1 << 3) -#define S3C2412_IISMOD_RCLK_384FS (2 << 3) -#define S3C2412_IISMOD_RCLK_768FS (3 << 3) -#define S3C2412_IISMOD_RCLK_MASK (3 << 3) -#define S3C2412_IISMOD_BCLK_32FS (0 << 1) -#define S3C2412_IISMOD_BCLK_48FS (1 << 1) -#define S3C2412_IISMOD_BCLK_16FS (2 << 1) -#define S3C2412_IISMOD_BCLK_24FS (3 << 1) -#define S3C2412_IISMOD_BCLK_MASK (3 << 1) -#define S3C2412_IISMOD_8BIT (1 << 0) - -#define S3C64XX_IISMOD_CDCLKCON (1 << 12) - -#define S3C2412_IISPSR_PSREN (1 << 15) - -#define S3C64XX_IISFIC_TX2COUNT(x) (((x) >> 24) & 0xf) -#define S3C64XX_IISFIC_TX1COUNT(x) (((x) >> 16) & 0xf) - -#define S3C2412_IISFIC_TXFLUSH (1 << 15) -#define S3C2412_IISFIC_RXFLUSH (1 << 7) -#define S3C2412_IISFIC_TXCOUNT(x) (((x) >> 8) & 0xf) -#define S3C2412_IISFIC_RXCOUNT(x) (((x) >> 0) & 0xf) - -#define S5PC1XX_IISFICS_TXFLUSH (1 << 15) -#define S5PC1XX_IISFICS_TXCOUNT(x) (((x) >> 8) & 0x7f) - -#endif /* __ASM_ARCH_REGS_S3C2412_IIS_H */ diff --git a/sound/soc/s3c24xx/s3c-ac97.c b/sound/soc/s3c24xx/s3c-ac97.c deleted file mode 100644 index ecf4fd0..0000000 --- a/sound/soc/s3c24xx/s3c-ac97.c +++ /dev/null @@ -1,521 +0,0 @@ -/* sound/soc/s3c24xx/s3c-ac97.c - * - * ALSA SoC Audio Layer - S3C AC97 Controller driver - * Evolved from s3c2443-ac97.c - * - * Copyright (c) 2010 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.com> - * Credits: Graeme Gregory, Sean Choi - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/io.h> -#include <linux/delay.h> -#include <linux/clk.h> - -#include <sound/soc.h> - -#include <plat/regs-ac97.h> -#include <mach/dma.h> -#include <plat/audio.h> - -#include "s3c-dma.h" -#include "s3c-ac97.h" - -#define AC_CMD_ADDR(x) (x << 16) -#define AC_CMD_DATA(x) (x & 0xffff) - -struct s3c_ac97_info { - unsigned state; - struct clk *ac97_clk; - void __iomem *regs; - struct mutex lock; - struct completion done; -}; -static struct s3c_ac97_info s3c_ac97; - -static struct s3c2410_dma_client s3c_dma_client_out = { - .name = "AC97 PCMOut" -}; - -static struct s3c2410_dma_client s3c_dma_client_in = { - .name = "AC97 PCMIn" -}; - -static struct s3c2410_dma_client s3c_dma_client_micin = { - .name = "AC97 MicIn" -}; - -static struct s3c_dma_params s3c_ac97_pcm_out = { - .client = &s3c_dma_client_out, - .dma_size = 4, -}; - -static struct s3c_dma_params s3c_ac97_pcm_in = { - .client = &s3c_dma_client_in, - .dma_size = 4, -}; - -static struct s3c_dma_params s3c_ac97_mic_in = { - .client = &s3c_dma_client_micin, - .dma_size = 4, -}; - -static void s3c_ac97_activate(struct snd_ac97 *ac97) -{ - u32 ac_glbctrl, stat; - - stat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT) & 0x7; - if (stat == S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE) - return; /* Return if already active */ - - INIT_COMPLETION(s3c_ac97.done); - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON; - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) - printk(KERN_ERR "AC97: Unable to activate!"); -} - -static unsigned short s3c_ac97_read(struct snd_ac97 *ac97, - unsigned short reg) -{ - u32 ac_glbctrl, ac_codec_cmd; - u32 stat, addr, data; - - mutex_lock(&s3c_ac97.lock); - - s3c_ac97_activate(ac97); - - INIT_COMPLETION(s3c_ac97.done); - - ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); - ac_codec_cmd = S3C_AC97_CODEC_CMD_READ | AC_CMD_ADDR(reg); - writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); - - udelay(50); - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) - printk(KERN_ERR "AC97: Unable to read!"); - - stat = readl(s3c_ac97.regs + S3C_AC97_STAT); - addr = (stat >> 16) & 0x7f; - data = (stat & 0xffff); - - if (addr != reg) - printk(KERN_ERR "s3c-ac97: req addr = %02x, rep addr = %02x\n", reg, addr); - - mutex_unlock(&s3c_ac97.lock); - - return (unsigned short)data; -} - -static void s3c_ac97_write(struct snd_ac97 *ac97, unsigned short reg, - unsigned short val) -{ - u32 ac_glbctrl, ac_codec_cmd; - - mutex_lock(&s3c_ac97.lock); - - s3c_ac97_activate(ac97); - - INIT_COMPLETION(s3c_ac97.done); - - ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); - ac_codec_cmd = AC_CMD_ADDR(reg) | AC_CMD_DATA(val); - writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); - - udelay(50); - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) - printk(KERN_ERR "AC97: Unable to write!"); - - ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); - ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; - writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); - - mutex_unlock(&s3c_ac97.lock); -} - -static void s3c_ac97_cold_reset(struct snd_ac97 *ac97) -{ - writel(S3C_AC97_GLBCTRL_COLDRESET, - s3c_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - writel(0, s3c_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); -} - -static void s3c_ac97_warm_reset(struct snd_ac97 *ac97) -{ - u32 stat; - - stat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT) & 0x7; - if (stat == S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE) - return; /* Return if already active */ - - writel(S3C_AC97_GLBCTRL_WARMRESET, s3c_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - writel(0, s3c_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - s3c_ac97_activate(ac97); -} - -static irqreturn_t s3c_ac97_irq(int irq, void *dev_id) -{ - u32 ac_glbctrl, ac_glbstat; - - ac_glbstat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT); - - if (ac_glbstat & S3C_AC97_GLBSTAT_CODECREADY) { - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl &= ~S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - complete(&s3c_ac97.done); - } - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl |= (1<<30); /* Clear interrupt */ - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - return IRQ_HANDLED; -} - -struct snd_ac97_bus_ops soc_ac97_ops = { - .read = s3c_ac97_read, - .write = s3c_ac97_write, - .warm_reset = s3c_ac97_warm_reset, - .reset = s3c_ac97_cold_reset, -}; -EXPORT_SYMBOL_GPL(soc_ac97_ops); - -static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - struct s3c_dma_params *dma_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c_ac97_pcm_out; - else - dma_data = &s3c_ac97_pcm_in; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - return 0; -} - -static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK; - else - ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMOUTTM_MASK; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA; - else - ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA; - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - break; - } - - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); - - return 0; -} - -static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); - - return 0; -} - -static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ac_glbctrl |= S3C_AC97_GLBCTRL_MICINTM_DMA; - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - break; - } - - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); - - return 0; -} - -static struct snd_soc_dai_ops s3c_ac97_dai_ops = { - .hw_params = s3c_ac97_hw_params, - .trigger = s3c_ac97_trigger, -}; - -static struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { - .hw_params = s3c_ac97_hw_mic_params, - .trigger = s3c_ac97_mic_trigger, -}; - -struct snd_soc_dai s3c_ac97_dai[] = { - [S3C_AC97_DAI_PCM] = { - .name = "s3c-ac97", - .id = S3C_AC97_DAI_PCM, - .ac97_control = 1, - .playback = { - .stream_name = "AC97 Playback", - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .stream_name = "AC97 Capture", - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &s3c_ac97_dai_ops, - }, - [S3C_AC97_DAI_MIC] = { - .name = "s3c-ac97-mic", - .id = S3C_AC97_DAI_MIC, - .ac97_control = 1, - .capture = { - .stream_name = "AC97 Mic Capture", - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &s3c_ac97_mic_dai_ops, - }, -}; -EXPORT_SYMBOL_GPL(s3c_ac97_dai); - -static __devinit int s3c_ac97_probe(struct platform_device *pdev) -{ - struct resource *mem_res, *dmatx_res, *dmarx_res, *dmamic_res, *irq_res; - struct s3c_audio_pdata *ac97_pdata; - int ret; - - ac97_pdata = pdev->dev.platform_data; - if (!ac97_pdata || !ac97_pdata->cfg_gpio) { - dev_err(&pdev->dev, "cfg_gpio callback not provided!\n"); - return -EINVAL; - } - - /* Check for availability of necessary resource */ - dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmatx_res) { - dev_err(&pdev->dev, "Unable to get AC97-TX dma resource\n"); - return -ENXIO; - } - - dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!dmarx_res) { - dev_err(&pdev->dev, "Unable to get AC97-RX dma resource\n"); - return -ENXIO; - } - - dmamic_res = platform_get_resource(pdev, IORESOURCE_DMA, 2); - if (!dmamic_res) { - dev_err(&pdev->dev, "Unable to get AC97-MIC dma resource\n"); - return -ENXIO; - } - - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem_res) { - dev_err(&pdev->dev, "Unable to get register resource\n"); - return -ENXIO; - } - - irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!irq_res) { - dev_err(&pdev->dev, "AC97 IRQ not provided!\n"); - return -ENXIO; - } - - if (!request_mem_region(mem_res->start, - resource_size(mem_res), "s3c-ac97")) { - dev_err(&pdev->dev, "Unable to request register region\n"); - return -EBUSY; - } - - s3c_ac97_pcm_out.channel = dmatx_res->start; - s3c_ac97_pcm_out.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; - s3c_ac97_pcm_in.channel = dmarx_res->start; - s3c_ac97_pcm_in.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; - s3c_ac97_mic_in.channel = dmamic_res->start; - s3c_ac97_mic_in.dma_addr = mem_res->start + S3C_AC97_MIC_DATA; - - init_completion(&s3c_ac97.done); - mutex_init(&s3c_ac97.lock); - - s3c_ac97.regs = ioremap(mem_res->start, resource_size(mem_res)); - if (s3c_ac97.regs == NULL) { - dev_err(&pdev->dev, "Unable to ioremap register region\n"); - ret = -ENXIO; - goto err1; - } - - s3c_ac97.ac97_clk = clk_get(&pdev->dev, "ac97"); - if (IS_ERR(s3c_ac97.ac97_clk)) { - dev_err(&pdev->dev, "s3c-ac97 failed to get ac97_clock\n"); - ret = -ENODEV; - goto err2; - } - clk_enable(s3c_ac97.ac97_clk); - - if (ac97_pdata->cfg_gpio(pdev)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err3; - } - - ret = request_irq(irq_res->start, s3c_ac97_irq, - IRQF_DISABLED, "AC97", NULL); - if (ret < 0) { - printk(KERN_ERR "s3c-ac97: interrupt request failed.\n"); - goto err4; - } - - s3c_ac97_dai[S3C_AC97_DAI_PCM].dev = &pdev->dev; - s3c_ac97_dai[S3C_AC97_DAI_MIC].dev = &pdev->dev; - - ret = snd_soc_register_dais(s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); - if (ret) - goto err5; - - return 0; - -err5: - free_irq(irq_res->start, NULL); -err4: -err3: - clk_disable(s3c_ac97.ac97_clk); - clk_put(s3c_ac97.ac97_clk); -err2: - iounmap(s3c_ac97.regs); -err1: - release_mem_region(mem_res->start, resource_size(mem_res)); - - return ret; -} - -static __devexit int s3c_ac97_remove(struct platform_device *pdev) -{ - struct resource *mem_res, *irq_res; - - snd_soc_unregister_dais(s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); - - irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (irq_res) - free_irq(irq_res->start, NULL); - - clk_disable(s3c_ac97.ac97_clk); - clk_put(s3c_ac97.ac97_clk); - - iounmap(s3c_ac97.regs); - - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (mem_res) - release_mem_region(mem_res->start, resource_size(mem_res)); - - return 0; -} - -static struct platform_driver s3c_ac97_driver = { - .probe = s3c_ac97_probe, - .remove = s3c_ac97_remove, - .driver = { - .name = "s3c-ac97", - .owner = THIS_MODULE, - }, -}; - -static int __init s3c_ac97_init(void) -{ - return platform_driver_register(&s3c_ac97_driver); -} -module_init(s3c_ac97_init); - -static void __exit s3c_ac97_exit(void) -{ - platform_driver_unregister(&s3c_ac97_driver); -} -module_exit(s3c_ac97_exit); - -MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); -MODULE_DESCRIPTION("AC97 driver for the Samsung SoC"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-ac97.h b/sound/soc/s3c24xx/s3c-ac97.h deleted file mode 100644 index 2781983..0000000 --- a/sound/soc/s3c24xx/s3c-ac97.h +++ /dev/null @@ -1,23 +0,0 @@ -/* sound/soc/s3c24xx/s3c-ac97.h - * - * ALSA SoC Audio Layer - S3C AC97 Controller driver - * Evolved from s3c2443-ac97.h - * - * Copyright (c) 2010 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.com> - * Credits: Graeme Gregory, Sean Choi - * - * 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 __S3C_AC97_H_ -#define __S3C_AC97_H_ - -#define S3C_AC97_DAI_PCM 0 -#define S3C_AC97_DAI_MIC 1 - -extern struct snd_soc_dai s3c_ac97_dai[]; - -#endif /* __S3C_AC97_H_ */ diff --git a/sound/soc/s3c24xx/s3c-dma.c b/sound/soc/s3c24xx/s3c-dma.c deleted file mode 100644 index 1b61c23..0000000 --- a/sound/soc/s3c24xx/s3c-dma.c +++ /dev/null @@ -1,483 +0,0 @@ -/* - * s3c-dma.c -- ALSA Soc Audio Layer - * - * (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com - * - * Copyright 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/io.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/dma.h> -#include <mach/hardware.h> -#include <mach/dma.h> - -#include "s3c-dma.h" - -static const struct snd_pcm_hardware s3c_dma_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_U16_LE | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S8, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 128*1024, - .period_bytes_min = PAGE_SIZE, - .period_bytes_max = PAGE_SIZE*2, - .periods_min = 2, - .periods_max = 128, - .fifo_size = 32, -}; - -struct s3c24xx_runtime_data { - spinlock_t lock; - int state; - unsigned int dma_loaded; - unsigned int dma_limit; - unsigned int dma_period; - dma_addr_t dma_start; - dma_addr_t dma_pos; - dma_addr_t dma_end; - struct s3c_dma_params *params; -}; - -/* s3c_dma_enqueue - * - * place a dma buffer onto the queue for the dma system - * to handle. -*/ -static void s3c_dma_enqueue(struct snd_pcm_substream *substream) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - dma_addr_t pos = prtd->dma_pos; - unsigned int limit; - int ret; - - pr_debug("Entered %s\n", __func__); - - if (s3c_dma_has_circular()) - limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; - else - limit = prtd->dma_limit; - - pr_debug("%s: loaded %d, limit %d\n", - __func__, prtd->dma_loaded, limit); - - while (prtd->dma_loaded < limit) { - unsigned long len = prtd->dma_period; - - pr_debug("dma_loaded: %d\n", prtd->dma_loaded); - - if ((pos + len) > prtd->dma_end) { - len = prtd->dma_end - pos; - pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n", - __func__, len); - } - - ret = s3c2410_dma_enqueue(prtd->params->channel, - substream, pos, len); - - if (ret == 0) { - prtd->dma_loaded++; - pos += prtd->dma_period; - if (pos >= prtd->dma_end) - pos = prtd->dma_start; - } else - break; - } - - prtd->dma_pos = pos; -} - -static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, - void *dev_id, int size, - enum s3c2410_dma_buffresult result) -{ - struct snd_pcm_substream *substream = dev_id; - struct s3c24xx_runtime_data *prtd; - - pr_debug("Entered %s\n", __func__); - - if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR) - return; - - prtd = substream->runtime->private_data; - - if (substream) - snd_pcm_period_elapsed(substream); - - spin_lock(&prtd->lock); - if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { - prtd->dma_loaded--; - s3c_dma_enqueue(substream); - } - - spin_unlock(&prtd->lock); -} - -static int s3c_dma_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - unsigned long totbytes = params_buffer_bytes(params); - struct s3c_dma_params *dma = - snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); - int ret = 0; - - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!dma) - return 0; - - /* this may get called several times by oss emulation - * with different params -HW */ - if (prtd->params == NULL) { - /* prepare DMA */ - prtd->params = dma; - - pr_debug("params %p, client %p, channel %d\n", prtd->params, - prtd->params->client, prtd->params->channel); - - ret = s3c2410_dma_request(prtd->params->channel, - prtd->params->client, NULL); - - if (ret < 0) { - printk(KERN_ERR "failed to get dma channel\n"); - return ret; - } - - /* use the circular buffering if we have it available. */ - if (s3c_dma_has_circular()) - s3c2410_dma_setflags(prtd->params->channel, - S3C2410_DMAF_CIRCULAR); - } - - s3c2410_dma_set_buffdone_fn(prtd->params->channel, - s3c24xx_audio_buffdone); - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - runtime->dma_bytes = totbytes; - - spin_lock_irq(&prtd->lock); - prtd->dma_loaded = 0; - prtd->dma_limit = runtime->hw.periods_min; - prtd->dma_period = params_period_bytes(params); - prtd->dma_start = runtime->dma_addr; - prtd->dma_pos = prtd->dma_start; - prtd->dma_end = prtd->dma_start + totbytes; - spin_unlock_irq(&prtd->lock); - - return 0; -} - -static int s3c_dma_hw_free(struct snd_pcm_substream *substream) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - /* TODO - do we need to ensure DMA flushed */ - snd_pcm_set_runtime_buffer(substream, NULL); - - if (prtd->params) { - s3c2410_dma_free(prtd->params->channel, prtd->params->client); - prtd->params = NULL; - } - - return 0; -} - -static int s3c_dma_prepare(struct snd_pcm_substream *substream) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!prtd->params) - return 0; - - /* channel needs configuring for mem=>device, increment memory addr, - * sync to pclk, half-word transfers to the IIS-FIFO. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - s3c2410_dma_devconfig(prtd->params->channel, - S3C2410_DMASRC_MEM, - prtd->params->dma_addr); - } else { - s3c2410_dma_devconfig(prtd->params->channel, - S3C2410_DMASRC_HW, - prtd->params->dma_addr); - } - - s3c2410_dma_config(prtd->params->channel, - prtd->params->dma_size); - - /* flush the DMA channel */ - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH); - prtd->dma_loaded = 0; - prtd->dma_pos = prtd->dma_start; - - /* enqueue dma buffers */ - s3c_dma_enqueue(substream); - - return ret; -} - -static int s3c_dma_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - spin_lock(&prtd->lock); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - prtd->state |= ST_RUNNING; - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - prtd->state &= ~ST_RUNNING; - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STOP); - break; - - default: - ret = -EINVAL; - break; - } - - spin_unlock(&prtd->lock); - - return ret; -} - -static snd_pcm_uframes_t -s3c_dma_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd = runtime->private_data; - unsigned long res; - dma_addr_t src, dst; - - pr_debug("Entered %s\n", __func__); - - spin_lock(&prtd->lock); - s3c2410_dma_getposition(prtd->params->channel, &src, &dst); - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - res = dst - prtd->dma_start; - else - res = src - prtd->dma_start; - - spin_unlock(&prtd->lock); - - pr_debug("Pointer %x %x\n", src, dst); - - /* we seem to be getting the odd error from the pcm library due - * to out-of-bounds pointers. this is maybe due to the dma engine - * not having loaded the new values for the channel before being - * callled... (todo - fix ) - */ - - if (res >= snd_pcm_lib_buffer_bytes(substream)) { - if (res == snd_pcm_lib_buffer_bytes(substream)) - res = 0; - } - - return bytes_to_frames(substream->runtime, res); -} - -static int s3c_dma_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd; - - pr_debug("Entered %s\n", __func__); - - snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - snd_soc_set_runtime_hwparams(substream, &s3c_dma_hardware); - - prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - spin_lock_init(&prtd->lock); - - runtime->private_data = prtd; - return 0; -} - -static int s3c_dma_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd = runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - if (!prtd) - pr_debug("s3c_dma_close called with prtd == NULL\n"); - - kfree(prtd); - - return 0; -} - -static int s3c_dma_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - pr_debug("Entered %s\n", __func__); - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops s3c_dma_ops = { - .open = s3c_dma_open, - .close = s3c_dma_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = s3c_dma_hw_params, - .hw_free = s3c_dma_hw_free, - .prepare = s3c_dma_prepare, - .trigger = s3c_dma_trigger, - .pointer = s3c_dma_pointer, - .mmap = s3c_dma_mmap, -}; - -static int s3c_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = s3c_dma_hardware.buffer_bytes_max; - - pr_debug("Entered %s\n", __func__); - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - buf->bytes = size; - return 0; -} - -static void s3c_dma_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - pr_debug("Entered %s\n", __func__); - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -static u64 s3c_dma_mask = DMA_BIT_MASK(32); - -static int s3c_dma_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) -{ - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - if (!card->dev->dma_mask) - card->dev->dma_mask = &s3c_dma_mask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->playback.channels_min) { - ret = s3c_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - ret = s3c_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - -struct snd_soc_platform s3c24xx_soc_platform = { - .name = "s3c24xx-audio", - .pcm_ops = &s3c_dma_ops, - .pcm_new = s3c_dma_new, - .pcm_free = s3c_dma_free_dma_buffers, -}; -EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); - -static int __init s3c24xx_soc_platform_init(void) -{ - return snd_soc_register_platform(&s3c24xx_soc_platform); -} -module_init(s3c24xx_soc_platform_init); - -static void __exit s3c24xx_soc_platform_exit(void) -{ - snd_soc_unregister_platform(&s3c24xx_soc_platform); -} -module_exit(s3c24xx_soc_platform_exit); - -MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-dma.h b/sound/soc/s3c24xx/s3c-dma.h deleted file mode 100644 index 69bb6bf..0000000 --- a/sound/soc/s3c24xx/s3c-dma.h +++ /dev/null @@ -1,31 +0,0 @@ -/* - * s3c-dma.h -- - * - * 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. - * - * ALSA PCM interface for the Samsung S3C24xx CPU - */ - -#ifndef _S3C_AUDIO_H -#define _S3C_AUDIO_H - -#define ST_RUNNING (1<<0) -#define ST_OPENED (1<<1) - -struct s3c_dma_params { - struct s3c2410_dma_client *client; /* stream identifier */ - int channel; /* Channel ID */ - dma_addr_t dma_addr; - int dma_size; /* Size of the DMA transfer */ -}; - -#define S3C24XX_DAI_I2S 0 - -/* platform data */ -extern struct snd_soc_platform s3c24xx_soc_platform; -extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; - -#endif diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c deleted file mode 100644 index 13311c8..0000000 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ /dev/null @@ -1,776 +0,0 @@ -/* sound/soc/s3c24xx/s3c-i2c-v2.c - * - * ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. - * - * Copyright (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory(a)wolfsonmicro.com - * linux(a)wolfsonmicro.com - * - * Copyright (c) 2008, 2007, 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/io.h> - -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <mach/dma.h> - -#include "regs-i2s-v2.h" -#include "s3c-i2s-v2.h" -#include "s3c-dma.h" - -#undef S3C_IIS_V2_SUPPORTED - -#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) -#define S3C_IIS_V2_SUPPORTED -#endif - -#ifdef CONFIG_PLAT_S3C64XX -#define S3C_IIS_V2_SUPPORTED -#endif - -#ifndef S3C_IIS_V2_SUPPORTED -#error Unsupported CPU model -#endif - -#define S3C2412_I2S_DEBUG_CON 0 - -static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) -{ - return cpu_dai->private_data; -} - -#define bit_set(v, b) (((v) & (b)) ? 1 : 0) - -#if S3C2412_I2S_DEBUG_CON -static void dbg_showcon(const char *fn, u32 con) -{ - printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn, - bit_set(con, S3C2412_IISCON_LRINDEX), - bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY), - bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY), - bit_set(con, S3C2412_IISCON_TXFIFO_FULL), - bit_set(con, S3C2412_IISCON_RXFIFO_FULL)); - - printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n", - fn, - bit_set(con, S3C2412_IISCON_TXDMA_PAUSE), - bit_set(con, S3C2412_IISCON_RXDMA_PAUSE), - bit_set(con, S3C2412_IISCON_TXCH_PAUSE), - bit_set(con, S3C2412_IISCON_RXCH_PAUSE)); - printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn, - bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE), - bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE), - bit_set(con, S3C2412_IISCON_IIS_ACTIVE)); -} -#else -static inline void dbg_showcon(const char *fn, u32 con) -{ -} -#endif - - -/* Turn on or off the transmission path. */ -static void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) -{ - void __iomem *regs = i2s->regs; - u32 fic, con, mod; - - pr_debug("%s(%d)\n", __func__, on); - - fic = readl(regs + S3C2412_IISFIC); - con = readl(regs + S3C2412_IISCON); - mod = readl(regs + S3C2412_IISMOD); - - pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); - - if (on) { - con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; - con &= ~S3C2412_IISCON_TXDMA_PAUSE; - con &= ~S3C2412_IISCON_TXCH_PAUSE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_TXONLY: - case S3C2412_IISMOD_MODE_TXRX: - /* do nothing, we are in the right mode */ - break; - - case S3C2412_IISMOD_MODE_RXONLY: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_TXRX; - break; - - default: - dev_err(i2s->dev, "TXEN: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); - break; - } - - writel(con, regs + S3C2412_IISCON); - writel(mod, regs + S3C2412_IISMOD); - } else { - /* Note, we do not have any indication that the FIFO problems - * tha the S3C2410/2440 had apply here, so we should be able - * to disable the DMA and TX without resetting the FIFOS. - */ - - con |= S3C2412_IISCON_TXDMA_PAUSE; - con |= S3C2412_IISCON_TXCH_PAUSE; - con &= ~S3C2412_IISCON_TXDMA_ACTIVE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_TXRX: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_RXONLY; - break; - - case S3C2412_IISMOD_MODE_TXONLY: - mod &= ~S3C2412_IISMOD_MODE_MASK; - con &= ~S3C2412_IISCON_IIS_ACTIVE; - break; - - default: - dev_err(i2s->dev, "TXDIS: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); - break; - } - - writel(mod, regs + S3C2412_IISMOD); - writel(con, regs + S3C2412_IISCON); - } - - fic = readl(regs + S3C2412_IISFIC); - dbg_showcon(__func__, con); - pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); -} - -static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) -{ - void __iomem *regs = i2s->regs; - u32 fic, con, mod; - - pr_debug("%s(%d)\n", __func__, on); - - fic = readl(regs + S3C2412_IISFIC); - con = readl(regs + S3C2412_IISCON); - mod = readl(regs + S3C2412_IISMOD); - - pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); - - if (on) { - con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; - con &= ~S3C2412_IISCON_RXDMA_PAUSE; - con &= ~S3C2412_IISCON_RXCH_PAUSE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_TXRX: - case S3C2412_IISMOD_MODE_RXONLY: - /* do nothing, we are in the right mode */ - break; - - case S3C2412_IISMOD_MODE_TXONLY: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_TXRX; - break; - - default: - dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); - } - - writel(mod, regs + S3C2412_IISMOD); - writel(con, regs + S3C2412_IISCON); - } else { - /* See txctrl notes on FIFOs. */ - - con &= ~S3C2412_IISCON_RXDMA_ACTIVE; - con |= S3C2412_IISCON_RXDMA_PAUSE; - con |= S3C2412_IISCON_RXCH_PAUSE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_RXONLY: - con &= ~S3C2412_IISCON_IIS_ACTIVE; - mod &= ~S3C2412_IISMOD_MODE_MASK; - break; - - case S3C2412_IISMOD_MODE_TXRX: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_TXONLY; - break; - - default: - dev_err(i2s->dev, "RXDIS: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); - } - - writel(con, regs + S3C2412_IISCON); - writel(mod, regs + S3C2412_IISMOD); - } - - fic = readl(regs + S3C2412_IISFIC); - pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); -} - -#define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t) - -/* - * Wait for the LR signal to allow synchronisation to the L/R clock - * from the codec. May only be needed for slave mode. - */ -static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s) -{ - u32 iiscon; - unsigned long loops = msecs_to_loops(5); - - pr_debug("Entered %s\n", __func__); - - while (--loops) { - iiscon = readl(i2s->regs + S3C2412_IISCON); - if (iiscon & S3C2412_IISCON_LRINDEX) - break; - - cpu_relax(); - } - - if (!loops) { - printk(KERN_ERR "%s: timeout\n", __func__); - return -ETIMEDOUT; - } - - return 0; -} - -/* - * Set S3C2412 I2S DAI format - */ -static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - iismod = readl(i2s->regs + S3C2412_IISMOD); - pr_debug("hw_params r: IISMOD: %x \n", iismod); - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - i2s->master = 0; - iismod |= S3C2412_IISMOD_SLAVE; - break; - case SND_SOC_DAIFMT_CBS_CFS: - i2s->master = 1; - iismod &= ~S3C2412_IISMOD_SLAVE; - break; - default: - pr_err("unknwon master/slave format\n"); - return -EINVAL; - } - - iismod &= ~S3C2412_IISMOD_SDF_MASK; - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_RIGHT_J: - iismod |= S3C2412_IISMOD_LR_RLOW; - iismod |= S3C2412_IISMOD_SDF_MSB; - break; - case SND_SOC_DAIFMT_LEFT_J: - iismod |= S3C2412_IISMOD_LR_RLOW; - iismod |= S3C2412_IISMOD_SDF_LSB; - break; - case SND_SOC_DAIFMT_I2S: - iismod &= ~S3C2412_IISMOD_LR_RLOW; - iismod |= S3C2412_IISMOD_SDF_IIS; - break; - default: - pr_err("Unknown data format\n"); - return -EINVAL; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - pr_debug("hw_params w: IISMOD: %x \n", iismod); - return 0; -} - -static int s3c_i2sv2_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *socdai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai_link *dai = rtd->dai; - struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai); - struct s3c_dma_params *dma_data; - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = i2s->dma_playback; - else - dma_data = i2s->dma_capture; - - snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); - - /* Working copies of register */ - iismod = readl(i2s->regs + S3C2412_IISMOD); - pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); - - iismod &= ~S3C64XX_IISMOD_BLC_MASK; - /* Sample size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - iismod |= S3C64XX_IISMOD_BLC_8BIT; - break; - case SNDRV_PCM_FORMAT_S16_LE: - break; - case SNDRV_PCM_FORMAT_S24_LE: - iismod |= S3C64XX_IISMOD_BLC_24BIT; - break; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); - - return 0; -} - -static int s3c_i2sv2_set_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - - pr_debug("Entered %s\n", __func__); - pr_debug("%s r: IISMOD: %x\n", __func__, iismod); - - switch (clk_id) { - case S3C_I2SV2_CLKSRC_PCLK: - iismod &= ~S3C2412_IISMOD_IMS_SYSMUX; - break; - - case S3C_I2SV2_CLKSRC_AUDIOBUS: - iismod |= S3C2412_IISMOD_IMS_SYSMUX; - break; - - case S3C_I2SV2_CLKSRC_CDCLK: - /* Error if controller doesn't have the CDCLKCON bit */ - if (!(i2s->feature & S3C_FEATURE_CDCLKCON)) - return -EINVAL; - - switch (dir) { - case SND_SOC_CLOCK_IN: - iismod |= S3C64XX_IISMOD_CDCLKCON; - break; - case SND_SOC_CLOCK_OUT: - iismod &= ~S3C64XX_IISMOD_CDCLKCON; - break; - default: - return -EINVAL; - } - break; - - default: - return -EINVAL; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - pr_debug("%s w: IISMOD: %x\n", __func__, iismod); - - return 0; -} - -static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_i2sv2_info *i2s = to_info(rtd->dai->cpu_dai); - int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); - unsigned long irqs; - int ret = 0; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); - - pr_debug("Entered %s\n", __func__); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - /* On start, ensure that the FIFOs are cleared and reset. */ - - writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH, - i2s->regs + S3C2412_IISFIC); - - /* clear again, just in case */ - writel(0x0, i2s->regs + S3C2412_IISFIC); - - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!i2s->master) { - ret = s3c2412_snd_lrsync(i2s); - if (ret) - goto exit_err; - } - - local_irq_save(irqs); - - if (capture) - s3c2412_snd_rxctrl(i2s, 1); - else - s3c2412_snd_txctrl(i2s, 1); - - local_irq_restore(irqs); - - /* - * Load the next buffer to DMA to meet the reqirement - * of the auto reload mechanism of S3C24XX. - * This call won't bother S3C64XX. - */ - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); - - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - local_irq_save(irqs); - - if (capture) - s3c2412_snd_rxctrl(i2s, 0); - else - s3c2412_snd_txctrl(i2s, 0); - - local_irq_restore(irqs); - break; - default: - ret = -EINVAL; - break; - } - -exit_err: - return ret; -} - -/* - * Set S3C2412 Clock dividers - */ -static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - u32 reg; - - pr_debug("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div); - - switch (div_id) { - case S3C_I2SV2_DIV_BCLK: - switch (div) { - case 16: - div = S3C2412_IISMOD_BCLK_16FS; - break; - - case 32: - div = S3C2412_IISMOD_BCLK_32FS; - break; - - case 24: - div = S3C2412_IISMOD_BCLK_24FS; - break; - - case 48: - div = S3C2412_IISMOD_BCLK_48FS; - break; - - default: - return -EINVAL; - } - - reg = readl(i2s->regs + S3C2412_IISMOD); - reg &= ~S3C2412_IISMOD_BCLK_MASK; - writel(reg | div, i2s->regs + S3C2412_IISMOD); - - pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); - break; - - case S3C_I2SV2_DIV_RCLK: - switch (div) { - case 256: - div = S3C2412_IISMOD_RCLK_256FS; - break; - - case 384: - div = S3C2412_IISMOD_RCLK_384FS; - break; - - case 512: - div = S3C2412_IISMOD_RCLK_512FS; - break; - - case 768: - div = S3C2412_IISMOD_RCLK_768FS; - break; - - default: - return -EINVAL; - } - - reg = readl(i2s->regs + S3C2412_IISMOD); - reg &= ~S3C2412_IISMOD_RCLK_MASK; - writel(reg | div, i2s->regs + S3C2412_IISMOD); - pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); - break; - - case S3C_I2SV2_DIV_PRESCALER: - if (div >= 0) { - writel((div << 8) | S3C2412_IISPSR_PSREN, - i2s->regs + S3C2412_IISPSR); - } else { - writel(0x0, i2s->regs + S3C2412_IISPSR); - } - pr_debug("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR)); - break; - - default: - return -EINVAL; - } - - return 0; -} - -static snd_pcm_sframes_t s3c2412_i2s_delay(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - u32 reg = readl(i2s->regs + S3C2412_IISFIC); - snd_pcm_sframes_t delay; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - delay = S3C2412_IISFIC_TXCOUNT(reg); - else - delay = S3C2412_IISFIC_RXCOUNT(reg); - - return delay; -} - -struct clk *s3c_i2sv2_get_clock(struct snd_soc_dai *cpu_dai) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - - if (iismod & S3C2412_IISMOD_IMS_SYSMUX) - return i2s->iis_cclk; - else - return i2s->iis_pclk; -} -EXPORT_SYMBOL_GPL(s3c_i2sv2_get_clock); - -/* default table of all avaialable root fs divisors */ -static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 }; - -int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, - unsigned int *fstab, - unsigned int rate, struct clk *clk) -{ - unsigned long clkrate = clk_get_rate(clk); - unsigned int div; - unsigned int fsclk; - unsigned int actual; - unsigned int fs; - unsigned int fsdiv; - signed int deviation = 0; - unsigned int best_fs = 0; - unsigned int best_div = 0; - unsigned int best_rate = 0; - unsigned int best_deviation = INT_MAX; - - pr_debug("Input clock rate %ldHz\n", clkrate); - - if (fstab == NULL) - fstab = iis_fs_tab; - - for (fs = 0; fs < ARRAY_SIZE(iis_fs_tab); fs++) { - fsdiv = iis_fs_tab[fs]; - - fsclk = clkrate / fsdiv; - div = fsclk / rate; - - if ((fsclk % rate) > (rate / 2)) - div++; - - if (div <= 1) - continue; - - actual = clkrate / (fsdiv * div); - deviation = actual - rate; - - printk(KERN_DEBUG "%ufs: div %u => result %u, deviation %d\n", - fsdiv, div, actual, deviation); - - deviation = abs(deviation); - - if (deviation < best_deviation) { - best_fs = fsdiv; - best_div = div; - best_rate = actual; - best_deviation = deviation; - } - - if (deviation == 0) - break; - } - - printk(KERN_DEBUG "best: fs=%u, div=%u, rate=%u\n", - best_fs, best_div, best_rate); - - info->fs_div = best_fs; - info->clk_div = best_div; - - return 0; -} -EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate); - -int s3c_i2sv2_probe(struct platform_device *pdev, - struct snd_soc_dai *dai, - struct s3c_i2sv2_info *i2s, - unsigned long base) -{ - struct device *dev = &pdev->dev; - unsigned int iismod; - - i2s->dev = dev; - - /* record our i2s structure for later use in the callbacks */ - dai->private_data = i2s; - - if (!base) { - struct resource *res = platform_get_resource(pdev, - IORESOURCE_MEM, - 0); - if (!res) { - dev_err(dev, "Unable to get register resource\n"); - return -ENXIO; - } - - if (!request_mem_region(res->start, resource_size(res), - "s3c64xx-i2s-v4")) { - dev_err(dev, "Unable to request register region\n"); - return -EBUSY; - } - - base = res->start; - } - - i2s->regs = ioremap(base, 0x100); - if (i2s->regs == NULL) { - dev_err(dev, "cannot ioremap registers\n"); - return -ENXIO; - } - - i2s->iis_pclk = clk_get(dev, "iis"); - if (IS_ERR(i2s->iis_pclk)) { - dev_err(dev, "failed to get iis_clock\n"); - iounmap(i2s->regs); - return -ENOENT; - } - - clk_enable(i2s->iis_pclk); - - /* Mark ourselves as in TXRX mode so we can run through our cleanup - * process without warnings. */ - iismod = readl(i2s->regs + S3C2412_IISMOD); - iismod |= S3C2412_IISMOD_MODE_TXRX; - writel(iismod, i2s->regs + S3C2412_IISMOD); - s3c2412_snd_txctrl(i2s, 0); - s3c2412_snd_rxctrl(i2s, 0); - - return 0; -} -EXPORT_SYMBOL_GPL(s3c_i2sv2_probe); - -#ifdef CONFIG_PM -static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - u32 iismod; - - if (dai->active) { - i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); - i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); - i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); - - /* some basic suspend checks */ - - iismod = readl(i2s->regs + S3C2412_IISMOD); - - if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) - pr_warning("%s: RXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) - pr_warning("%s: TXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_IIS_ACTIVE) - pr_warning("%s: IIS active\n", __func__); - } - - return 0; -} - -static int s3c2412_i2s_resume(struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - - pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n", - dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); - - if (dai->active) { - writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); - writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); - writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); - - writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, - i2s->regs + S3C2412_IISFIC); - - ndelay(250); - writel(0x0, i2s->regs + S3C2412_IISFIC); - } - - return 0; -} -#else -#define s3c2412_i2s_suspend NULL -#define s3c2412_i2s_resume NULL -#endif - -int s3c_i2sv2_register_dai(struct snd_soc_dai *dai) -{ - struct snd_soc_dai_ops *ops = dai->ops; - - ops->trigger = s3c2412_i2s_trigger; - if (!ops->hw_params) - ops->hw_params = s3c_i2sv2_hw_params; - ops->set_fmt = s3c2412_i2s_set_fmt; - ops->set_clkdiv = s3c2412_i2s_set_clkdiv; - ops->set_sysclk = s3c_i2sv2_set_sysclk; - - /* Allow overriding by (for example) IISv4 */ - if (!ops->delay) - ops->delay = s3c2412_i2s_delay; - - dai->suspend = s3c2412_i2s_suspend; - dai->resume = s3c2412_i2s_resume; - - return snd_soc_register_dai(dai); -} -EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai); - -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h deleted file mode 100644 index 766f43a..0000000 --- a/sound/soc/s3c24xx/s3c-i2s-v2.h +++ /dev/null @@ -1,103 +0,0 @@ -/* sound/soc/s3c24xx/s3c-i2s-v2.h - * - * ALSA Soc Audio Layer - S3C_I2SV2 I2S driver - * - * Copyright (c) 2007 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. -*/ - -/* This code is the core support for the I2S block found in a number of - * Samsung SoC devices which is unofficially named I2S-V2. Currently the - * S3C2412 and the S3C64XX series use this block to provide 1 or 2 I2S - * channels via configurable GPIO. - */ - -#ifndef __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H -#define __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H __FILE__ - -#define S3C_I2SV2_DIV_BCLK (1) -#define S3C_I2SV2_DIV_RCLK (2) -#define S3C_I2SV2_DIV_PRESCALER (3) - -#define S3C_I2SV2_CLKSRC_PCLK 0 -#define S3C_I2SV2_CLKSRC_AUDIOBUS 1 -#define S3C_I2SV2_CLKSRC_CDCLK 2 - -/* Set this flag for I2S controllers that have the bit IISMOD[12] - * bridge/break RCLK signal and external Xi2sCDCLK pin. - */ -#define S3C_FEATURE_CDCLKCON (1 << 0) - -/** - * struct s3c_i2sv2_info - S3C I2S-V2 information - * @dev: The parent device passed to use from the probe. - * @regs: The pointer to the device registe block. - * @feature: Set of bit-flags indicating features of the controller. - * @master: True if the I2S core is the I2S bit clock master. - * @dma_playback: DMA information for playback channel. - * @dma_capture: DMA information for capture channel. - * @suspend_iismod: PM save for the IISMOD register. - * @suspend_iiscon: PM save for the IISCON register. - * @suspend_iispsr: PM save for the IISPSR register. - * - * This is the private codec state for the hardware associated with an - * I2S channel such as the register mappings and clock sources. - */ -struct s3c_i2sv2_info { - struct device *dev; - void __iomem *regs; - - u32 feature; - - struct clk *iis_pclk; - struct clk *iis_cclk; - - unsigned char master; - - struct s3c_dma_params *dma_playback; - struct s3c_dma_params *dma_capture; - - u32 suspend_iismod; - u32 suspend_iiscon; - u32 suspend_iispsr; -}; - -extern struct clk *s3c_i2sv2_get_clock(struct snd_soc_dai *cpu_dai); - -struct s3c_i2sv2_rate_calc { - unsigned int clk_div; /* for prescaler */ - unsigned int fs_div; /* for root frame clock */ -}; - -extern int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, - unsigned int *fstab, - unsigned int rate, struct clk *clk); - -/** - * s3c_i2sv2_probe - probe for i2s device helper - * @pdev: The platform device supplied to the original probe. - * @dai: The ASoC DAI structure supplied to the original probe. - * @i2s: Our local i2s structure to fill in. - * @base: The base address for the registers. - */ -extern int s3c_i2sv2_probe(struct platform_device *pdev, - struct snd_soc_dai *dai, - struct s3c_i2sv2_info *i2s, - unsigned long base); - -/** - * s3c_i2sv2_register_dai - register dai with soc core - * @dai: The snd_soc_dai structure to register - * - * Fill in any missing fields and then register the given dai with the - * soc core. - */ -extern int s3c_i2sv2_register_dai(struct snd_soc_dai *dai); - -#endif /* __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H */ diff --git a/sound/soc/s3c24xx/s3c-pcm.c b/sound/soc/s3c24xx/s3c-pcm.c deleted file mode 100644 index 326f0a9..0000000 --- a/sound/soc/s3c24xx/s3c-pcm.c +++ /dev/null @@ -1,554 +0,0 @@ -/* sound/soc/s3c24xx/s3c-pcm.c - * - * ALSA SoC Audio Layer - S3C PCM-Controller driver - * - * Copyright (c) 2009 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.com> - * based upon I2S drivers by Ben Dooks. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/kernel.h> -#include <linux/gpio.h> -#include <linux/io.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <plat/audio.h> -#include <plat/dma.h> - -#include "s3c-dma.h" -#include "s3c-pcm.h" - -static struct s3c2410_dma_client s3c_pcm_dma_client_out = { - .name = "PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c_pcm_dma_client_in = { - .name = "PCM Stereo in" -}; - -static struct s3c_dma_params s3c_pcm_stereo_out[] = { - [0] = { - .client = &s3c_pcm_dma_client_out, - .dma_size = 4, - }, - [1] = { - .client = &s3c_pcm_dma_client_out, - .dma_size = 4, - }, -}; - -static struct s3c_dma_params s3c_pcm_stereo_in[] = { - [0] = { - .client = &s3c_pcm_dma_client_in, - .dma_size = 4, - }, - [1] = { - .client = &s3c_pcm_dma_client_in, - .dma_size = 4, - }, -}; - -static struct s3c_pcm_info s3c_pcm[2]; - -static inline struct s3c_pcm_info *to_info(struct snd_soc_dai *cpu_dai) -{ - return cpu_dai->private_data; -} - -static void s3c_pcm_snd_txctrl(struct s3c_pcm_info *pcm, int on) -{ - void __iomem *regs = pcm->regs; - u32 ctl, clkctl; - - clkctl = readl(regs + S3C_PCM_CLKCTL); - ctl = readl(regs + S3C_PCM_CTL); - ctl &= ~(S3C_PCM_CTL_TXDIPSTICK_MASK - << S3C_PCM_CTL_TXDIPSTICK_SHIFT); - - if (on) { - ctl |= S3C_PCM_CTL_TXDMA_EN; - ctl |= S3C_PCM_CTL_TXFIFO_EN; - ctl |= S3C_PCM_CTL_ENABLE; - ctl |= (0x20<<S3C_PCM_CTL_TXDIPSTICK_SHIFT); - clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; - } else { - ctl &= ~S3C_PCM_CTL_TXDMA_EN; - ctl &= ~S3C_PCM_CTL_TXFIFO_EN; - - if (!(ctl & S3C_PCM_CTL_RXFIFO_EN)) { - ctl &= ~S3C_PCM_CTL_ENABLE; - if (!pcm->idleclk) - clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; - } - } - - writel(clkctl, regs + S3C_PCM_CLKCTL); - writel(ctl, regs + S3C_PCM_CTL); -} - -static void s3c_pcm_snd_rxctrl(struct s3c_pcm_info *pcm, int on) -{ - void __iomem *regs = pcm->regs; - u32 ctl, clkctl; - - ctl = readl(regs + S3C_PCM_CTL); - clkctl = readl(regs + S3C_PCM_CLKCTL); - - if (on) { - ctl |= S3C_PCM_CTL_RXDMA_EN; - ctl |= S3C_PCM_CTL_RXFIFO_EN; - ctl |= S3C_PCM_CTL_ENABLE; - clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; - } else { - ctl &= ~S3C_PCM_CTL_RXDMA_EN; - ctl &= ~S3C_PCM_CTL_RXFIFO_EN; - - if (!(ctl & S3C_PCM_CTL_TXFIFO_EN)) { - ctl &= ~S3C_PCM_CTL_ENABLE; - if (!pcm->idleclk) - clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; - } - } - - writel(clkctl, regs + S3C_PCM_CLKCTL); - writel(ctl, regs + S3C_PCM_CTL); -} - -static int s3c_pcm_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_pcm_info *pcm = to_info(rtd->dai->cpu_dai); - unsigned long flags; - - dev_dbg(pcm->dev, "Entered %s\n", __func__); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - spin_lock_irqsave(&pcm->lock, flags); - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - s3c_pcm_snd_rxctrl(pcm, 1); - else - s3c_pcm_snd_txctrl(pcm, 1); - - spin_unlock_irqrestore(&pcm->lock, flags); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - spin_lock_irqsave(&pcm->lock, flags); - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - s3c_pcm_snd_rxctrl(pcm, 0); - else - s3c_pcm_snd_txctrl(pcm, 0); - - spin_unlock_irqrestore(&pcm->lock, flags); - break; - - default: - return -EINVAL; - } - - return 0; -} - -static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *socdai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai_link *dai = rtd->dai; - struct s3c_pcm_info *pcm = to_info(dai->cpu_dai); - struct s3c_dma_params *dma_data; - void __iomem *regs = pcm->regs; - struct clk *clk; - int sclk_div, sync_div; - unsigned long flags; - u32 clkctl; - - dev_dbg(pcm->dev, "Entered %s\n", __func__); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = pcm->dma_playback; - else - dma_data = pcm->dma_capture; - - snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); - - /* Strictly check for sample size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - break; - default: - return -EINVAL; - } - - spin_lock_irqsave(&pcm->lock, flags); - - /* Get hold of the PCMSOURCE_CLK */ - clkctl = readl(regs + S3C_PCM_CLKCTL); - if (clkctl & S3C_PCM_CLKCTL_SERCLKSEL_PCLK) - clk = pcm->pclk; - else - clk = pcm->cclk; - - /* Set the SCLK divider */ - sclk_div = clk_get_rate(clk) / pcm->sclk_per_fs / - params_rate(params) / 2 - 1; - - clkctl &= ~(S3C_PCM_CLKCTL_SCLKDIV_MASK - << S3C_PCM_CLKCTL_SCLKDIV_SHIFT); - clkctl |= ((sclk_div & S3C_PCM_CLKCTL_SCLKDIV_MASK) - << S3C_PCM_CLKCTL_SCLKDIV_SHIFT); - - /* Set the SYNC divider */ - sync_div = pcm->sclk_per_fs - 1; - - clkctl &= ~(S3C_PCM_CLKCTL_SYNCDIV_MASK - << S3C_PCM_CLKCTL_SYNCDIV_SHIFT); - clkctl |= ((sync_div & S3C_PCM_CLKCTL_SYNCDIV_MASK) - << S3C_PCM_CLKCTL_SYNCDIV_SHIFT); - - writel(clkctl, regs + S3C_PCM_CLKCTL); - - spin_unlock_irqrestore(&pcm->lock, flags); - - dev_dbg(pcm->dev, "PCMSOURCE_CLK-%lu SCLK=%ufs SCLK_DIV=%d SYNC_DIV=%d\n", - clk_get_rate(clk), pcm->sclk_per_fs, - sclk_div, sync_div); - - return 0; -} - -static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct s3c_pcm_info *pcm = to_info(cpu_dai); - void __iomem *regs = pcm->regs; - unsigned long flags; - int ret = 0; - u32 ctl; - - dev_dbg(pcm->dev, "Entered %s\n", __func__); - - spin_lock_irqsave(&pcm->lock, flags); - - ctl = readl(regs + S3C_PCM_CTL); - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - /* Nothing to do, NB_NF by default */ - break; - default: - dev_err(pcm->dev, "Unsupported clock inversion!\n"); - ret = -EINVAL; - goto exit; - } - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - /* Nothing to do, Master by default */ - break; - default: - dev_err(pcm->dev, "Unsupported master/slave format!\n"); - ret = -EINVAL; - goto exit; - } - - switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { - case SND_SOC_DAIFMT_CONT: - pcm->idleclk = 1; - break; - case SND_SOC_DAIFMT_GATED: - pcm->idleclk = 0; - break; - default: - dev_err(pcm->dev, "Invalid Clock gating request!\n"); - ret = -EINVAL; - goto exit; - } - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_DSP_A: - ctl |= S3C_PCM_CTL_TXMSB_AFTER_FSYNC; - ctl |= S3C_PCM_CTL_RXMSB_AFTER_FSYNC; - break; - case SND_SOC_DAIFMT_DSP_B: - ctl &= ~S3C_PCM_CTL_TXMSB_AFTER_FSYNC; - ctl &= ~S3C_PCM_CTL_RXMSB_AFTER_FSYNC; - break; - default: - dev_err(pcm->dev, "Unsupported data format!\n"); - ret = -EINVAL; - goto exit; - } - - writel(ctl, regs + S3C_PCM_CTL); - -exit: - spin_unlock_irqrestore(&pcm->lock, flags); - - return ret; -} - -static int s3c_pcm_set_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct s3c_pcm_info *pcm = to_info(cpu_dai); - - switch (div_id) { - case S3C_PCM_SCLK_PER_FS: - pcm->sclk_per_fs = div; - break; - - default: - return -EINVAL; - } - - return 0; -} - -static int s3c_pcm_set_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - struct s3c_pcm_info *pcm = to_info(cpu_dai); - void __iomem *regs = pcm->regs; - u32 clkctl = readl(regs + S3C_PCM_CLKCTL); - - switch (clk_id) { - case S3C_PCM_CLKSRC_PCLK: - clkctl |= S3C_PCM_CLKCTL_SERCLKSEL_PCLK; - break; - - case S3C_PCM_CLKSRC_MUX: - clkctl &= ~S3C_PCM_CLKCTL_SERCLKSEL_PCLK; - - if (clk_get_rate(pcm->cclk) != freq) - clk_set_rate(pcm->cclk, freq); - - break; - - default: - return -EINVAL; - } - - writel(clkctl, regs + S3C_PCM_CLKCTL); - - return 0; -} - -static struct snd_soc_dai_ops s3c_pcm_dai_ops = { - .set_sysclk = s3c_pcm_set_sysclk, - .set_clkdiv = s3c_pcm_set_clkdiv, - .trigger = s3c_pcm_trigger, - .hw_params = s3c_pcm_hw_params, - .set_fmt = s3c_pcm_set_fmt, -}; - -#define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 - -#define S3C_PCM_DECLARE(n) \ -{ \ - .name = "samsung-pcm", \ - .id = (n), \ - .symmetric_rates = 1, \ - .ops = &s3c_pcm_dai_ops, \ - .playback = { \ - .channels_min = 2, \ - .channels_max = 2, \ - .rates = S3C_PCM_RATES, \ - .formats = SNDRV_PCM_FMTBIT_S16_LE, \ - }, \ - .capture = { \ - .channels_min = 2, \ - .channels_max = 2, \ - .rates = S3C_PCM_RATES, \ - .formats = SNDRV_PCM_FMTBIT_S16_LE, \ - }, \ -} - -struct snd_soc_dai s3c_pcm_dai[] = { - S3C_PCM_DECLARE(0), - S3C_PCM_DECLARE(1), -}; -EXPORT_SYMBOL_GPL(s3c_pcm_dai); - -static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) -{ - struct s3c_pcm_info *pcm; - struct snd_soc_dai *dai; - struct resource *mem_res, *dmatx_res, *dmarx_res; - struct s3c_audio_pdata *pcm_pdata; - int ret; - - /* Check for valid device index */ - if ((pdev->id < 0) || pdev->id >= ARRAY_SIZE(s3c_pcm)) { - dev_err(&pdev->dev, "id %d out of range\n", pdev->id); - return -EINVAL; - } - - pcm_pdata = pdev->dev.platform_data; - - /* Check for availability of necessary resource */ - dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmatx_res) { - dev_err(&pdev->dev, "Unable to get PCM-TX dma resource\n"); - return -ENXIO; - } - - dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!dmarx_res) { - dev_err(&pdev->dev, "Unable to get PCM-RX dma resource\n"); - return -ENXIO; - } - - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem_res) { - dev_err(&pdev->dev, "Unable to get register resource\n"); - return -ENXIO; - } - - if (pcm_pdata && pcm_pdata->cfg_gpio && pcm_pdata->cfg_gpio(pdev)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - return -EINVAL; - } - - pcm = &s3c_pcm[pdev->id]; - pcm->dev = &pdev->dev; - - spin_lock_init(&pcm->lock); - - dai = &s3c_pcm_dai[pdev->id]; - dai->dev = &pdev->dev; - - /* Default is 128fs */ - pcm->sclk_per_fs = 128; - - pcm->cclk = clk_get(&pdev->dev, "audio-bus"); - if (IS_ERR(pcm->cclk)) { - dev_err(&pdev->dev, "failed to get audio-bus\n"); - ret = PTR_ERR(pcm->cclk); - goto err1; - } - clk_enable(pcm->cclk); - - /* record our pcm structure for later use in the callbacks */ - dai->private_data = pcm; - - if (!request_mem_region(mem_res->start, - resource_size(mem_res), "samsung-pcm")) { - dev_err(&pdev->dev, "Unable to request register region\n"); - ret = -EBUSY; - goto err2; - } - - pcm->regs = ioremap(mem_res->start, 0x100); - if (pcm->regs == NULL) { - dev_err(&pdev->dev, "cannot ioremap registers\n"); - ret = -ENXIO; - goto err3; - } - - pcm->pclk = clk_get(&pdev->dev, "pcm"); - if (IS_ERR(pcm->pclk)) { - dev_err(&pdev->dev, "failed to get pcm_clock\n"); - ret = -ENOENT; - goto err4; - } - clk_enable(pcm->pclk); - - ret = snd_soc_register_dai(dai); - if (ret != 0) { - dev_err(&pdev->dev, "failed to get pcm_clock\n"); - goto err5; - } - - s3c_pcm_stereo_in[pdev->id].dma_addr = mem_res->start - + S3C_PCM_RXFIFO; - s3c_pcm_stereo_out[pdev->id].dma_addr = mem_res->start - + S3C_PCM_TXFIFO; - - s3c_pcm_stereo_in[pdev->id].channel = dmarx_res->start; - s3c_pcm_stereo_out[pdev->id].channel = dmatx_res->start; - - pcm->dma_capture = &s3c_pcm_stereo_in[pdev->id]; - pcm->dma_playback = &s3c_pcm_stereo_out[pdev->id]; - - return 0; - -err5: - clk_disable(pcm->pclk); - clk_put(pcm->pclk); -err4: - iounmap(pcm->regs); -err3: - release_mem_region(mem_res->start, resource_size(mem_res)); -err2: - clk_disable(pcm->cclk); - clk_put(pcm->cclk); -err1: - return ret; -} - -static __devexit int s3c_pcm_dev_remove(struct platform_device *pdev) -{ - struct s3c_pcm_info *pcm = &s3c_pcm[pdev->id]; - struct resource *mem_res; - - iounmap(pcm->regs); - - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(mem_res->start, resource_size(mem_res)); - - clk_disable(pcm->cclk); - clk_disable(pcm->pclk); - clk_put(pcm->pclk); - clk_put(pcm->cclk); - - return 0; -} - -static struct platform_driver s3c_pcm_driver = { - .probe = s3c_pcm_dev_probe, - .remove = s3c_pcm_dev_remove, - .driver = { - .name = "samsung-pcm", - .owner = THIS_MODULE, - }, -}; - -static int __init s3c_pcm_init(void) -{ - return platform_driver_register(&s3c_pcm_driver); -} -module_init(s3c_pcm_init); - -static void __exit s3c_pcm_exit(void) -{ - platform_driver_unregister(&s3c_pcm_driver); -} -module_exit(s3c_pcm_exit); - -/* Module information */ -MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); -MODULE_DESCRIPTION("S3C PCM Controller Driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-pcm.h b/sound/soc/s3c24xx/s3c-pcm.h deleted file mode 100644 index 69ff997..0000000 --- a/sound/soc/s3c24xx/s3c-pcm.h +++ /dev/null @@ -1,123 +0,0 @@ -/* sound/soc/s3c24xx/s3c-pcm.h - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - */ - -#ifndef __S3C_PCM_H -#define __S3C_PCM_H __FILE__ - -/*Register Offsets */ -#define S3C_PCM_CTL (0x00) -#define S3C_PCM_CLKCTL (0x04) -#define S3C_PCM_TXFIFO (0x08) -#define S3C_PCM_RXFIFO (0x0C) -#define S3C_PCM_IRQCTL (0x10) -#define S3C_PCM_IRQSTAT (0x14) -#define S3C_PCM_FIFOSTAT (0x18) -#define S3C_PCM_CLRINT (0x20) - -/* PCM_CTL Bit-Fields */ -#define S3C_PCM_CTL_TXDIPSTICK_MASK (0x3f) -#define S3C_PCM_CTL_TXDIPSTICK_SHIFT (13) -#define S3C_PCM_CTL_RXDIPSTICK_MSK (0x3f<<7) -#define S3C_PCM_CTL_TXDMA_EN (0x1<<6) -#define S3C_PCM_CTL_RXDMA_EN (0x1<<5) -#define S3C_PCM_CTL_TXMSB_AFTER_FSYNC (0x1<<4) -#define S3C_PCM_CTL_RXMSB_AFTER_FSYNC (0x1<<3) -#define S3C_PCM_CTL_TXFIFO_EN (0x1<<2) -#define S3C_PCM_CTL_RXFIFO_EN (0x1<<1) -#define S3C_PCM_CTL_ENABLE (0x1<<0) - -/* PCM_CLKCTL Bit-Fields */ -#define S3C_PCM_CLKCTL_SERCLK_EN (0x1<<19) -#define S3C_PCM_CLKCTL_SERCLKSEL_PCLK (0x1<<18) -#define S3C_PCM_CLKCTL_SCLKDIV_MASK (0x1ff) -#define S3C_PCM_CLKCTL_SYNCDIV_MASK (0x1ff) -#define S3C_PCM_CLKCTL_SCLKDIV_SHIFT (9) -#define S3C_PCM_CLKCTL_SYNCDIV_SHIFT (0) - -/* PCM_TXFIFO Bit-Fields */ -#define S3C_PCM_TXFIFO_DVALID (0x1<<16) -#define S3C_PCM_TXFIFO_DATA_MSK (0xffff<<0) - -/* PCM_RXFIFO Bit-Fields */ -#define S3C_PCM_RXFIFO_DVALID (0x1<<16) -#define S3C_PCM_RXFIFO_DATA_MSK (0xffff<<0) - -/* PCM_IRQCTL Bit-Fields */ -#define S3C_PCM_IRQCTL_IRQEN (0x1<<14) -#define S3C_PCM_IRQCTL_WRDEN (0x1<<12) -#define S3C_PCM_IRQCTL_TXEMPTYEN (0x1<<11) -#define S3C_PCM_IRQCTL_TXALMSTEMPTYEN (0x1<<10) -#define S3C_PCM_IRQCTL_TXFULLEN (0x1<<9) -#define S3C_PCM_IRQCTL_TXALMSTFULLEN (0x1<<8) -#define S3C_PCM_IRQCTL_TXSTARVEN (0x1<<7) -#define S3C_PCM_IRQCTL_TXERROVRFLEN (0x1<<6) -#define S3C_PCM_IRQCTL_RXEMPTEN (0x1<<5) -#define S3C_PCM_IRQCTL_RXALMSTEMPTEN (0x1<<4) -#define S3C_PCM_IRQCTL_RXFULLEN (0x1<<3) -#define S3C_PCM_IRQCTL_RXALMSTFULLEN (0x1<<2) -#define S3C_PCM_IRQCTL_RXSTARVEN (0x1<<1) -#define S3C_PCM_IRQCTL_RXERROVRFLEN (0x1<<0) - -/* PCM_IRQSTAT Bit-Fields */ -#define S3C_PCM_IRQSTAT_IRQPND (0x1<<13) -#define S3C_PCM_IRQSTAT_WRD_XFER (0x1<<12) -#define S3C_PCM_IRQSTAT_TXEMPTY (0x1<<11) -#define S3C_PCM_IRQSTAT_TXALMSTEMPTY (0x1<<10) -#define S3C_PCM_IRQSTAT_TXFULL (0x1<<9) -#define S3C_PCM_IRQSTAT_TXALMSTFULL (0x1<<8) -#define S3C_PCM_IRQSTAT_TXSTARV (0x1<<7) -#define S3C_PCM_IRQSTAT_TXERROVRFL (0x1<<6) -#define S3C_PCM_IRQSTAT_RXEMPT (0x1<<5) -#define S3C_PCM_IRQSTAT_RXALMSTEMPT (0x1<<4) -#define S3C_PCM_IRQSTAT_RXFULL (0x1<<3) -#define S3C_PCM_IRQSTAT_RXALMSTFULL (0x1<<2) -#define S3C_PCM_IRQSTAT_RXSTARV (0x1<<1) -#define S3C_PCM_IRQSTAT_RXERROVRFL (0x1<<0) - -/* PCM_FIFOSTAT Bit-Fields */ -#define S3C_PCM_FIFOSTAT_TXCNT_MSK (0x3f<<14) -#define S3C_PCM_FIFOSTAT_TXFIFOEMPTY (0x1<<13) -#define S3C_PCM_FIFOSTAT_TXFIFOALMSTEMPTY (0x1<<12) -#define S3C_PCM_FIFOSTAT_TXFIFOFULL (0x1<<11) -#define S3C_PCM_FIFOSTAT_TXFIFOALMSTFULL (0x1<<10) -#define S3C_PCM_FIFOSTAT_RXCNT_MSK (0x3f<<4) -#define S3C_PCM_FIFOSTAT_RXFIFOEMPTY (0x1<<3) -#define S3C_PCM_FIFOSTAT_RXFIFOALMSTEMPTY (0x1<<2) -#define S3C_PCM_FIFOSTAT_RXFIFOFULL (0x1<<1) -#define S3C_PCM_FIFOSTAT_RXFIFOALMSTFULL (0x1<<0) - -#define S3C_PCM_CLKSRC_PCLK 0 -#define S3C_PCM_CLKSRC_MUX 1 - -#define S3C_PCM_SCLK_PER_FS 0 - -/** - * struct s3c_pcm_info - S3C PCM Controller information - * @dev: The parent device passed to use from the probe. - * @regs: The pointer to the device register block. - * @dma_playback: DMA information for playback channel. - * @dma_capture: DMA information for capture channel. - */ -struct s3c_pcm_info { - spinlock_t lock; - struct device *dev; - void __iomem *regs; - - unsigned int sclk_per_fs; - - /* Whether to keep PCMSCLK enabled even when idle(no active xfer) */ - unsigned int idleclk; - - struct clk *pclk; - struct clk *cclk; - - struct s3c_dma_params *dma_playback; - struct s3c_dma_params *dma_capture; -}; - -#endif /* __S3C_PCM_H */ diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c deleted file mode 100644 index 709adef..0000000 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ /dev/null @@ -1,190 +0,0 @@ -/* sound/soc/s3c24xx/s3c2412-i2s.c - * - * ALSA Soc Audio Layer - S3C2412 I2S driver - * - * Copyright (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory(a)wolfsonmicro.com - * linux(a)wolfsonmicro.com - * - * Copyright (c) 2007, 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/gpio.h> -#include <linux/clk.h> -#include <linux/kernel.h> -#include <linux/io.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> -#include <mach/hardware.h> - -#include <mach/regs-gpio.h> -#include <mach/dma.h> - -#include "s3c-dma.h" -#include "regs-i2s-v2.h" -#include "s3c2412-i2s.h" - -#define S3C2412_I2S_DEBUG 0 - -static struct s3c2410_dma_client s3c2412_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c2412_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - -static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = { - .client = &s3c2412_dma_client_out, - .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, - .dma_size = 4, -}; - -static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { - .client = &s3c2412_dma_client_in, - .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, - .dma_size = 4, -}; - -static struct s3c_i2sv2_info s3c2412_i2s; - -static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) -{ - return cpu_dai->private_data; -} - -static int s3c2412_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - int ret; - - pr_debug("Entered %s\n", __func__); - - ret = s3c_i2sv2_probe(pdev, dai, &s3c2412_i2s, S3C2410_PA_IIS); - if (ret) - return ret; - - s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in; - s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; - - s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk"); - if (s3c2412_i2s.iis_cclk == NULL) { - pr_err("failed to get i2sclk clock\n"); - iounmap(s3c2412_i2s.regs); - return -ENODEV; - } - - /* Set MPLL as the source for IIS CLK */ - - clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); - clk_enable(s3c2412_i2s.iis_cclk); - - s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk; - - /* Configure the I2S pins in correct mode */ - s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); - s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); - s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); - - return 0; -} - -static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - struct s3c_dma_params *dma_data; - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = i2s->dma_playback; - else - dma_data = i2s->dma_capture; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - iismod = readl(i2s->regs + S3C2412_IISMOD); - pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); - - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - iismod |= S3C2412_IISMOD_8BIT; - break; - case SNDRV_PCM_FORMAT_S16_LE: - iismod &= ~S3C2412_IISMOD_8BIT; - break; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); - - return 0; -} - -#define S3C2412_I2S_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - -static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = { - .hw_params = s3c2412_i2s_hw_params, -}; - -struct snd_soc_dai s3c2412_i2s_dai = { - .name = "s3c2412-i2s", - .id = 0, - .probe = s3c2412_i2s_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C2412_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C2412_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = &s3c2412_i2s_dai_ops, -}; -EXPORT_SYMBOL_GPL(s3c2412_i2s_dai); - -static int __init s3c2412_i2s_init(void) -{ - return s3c_i2sv2_register_dai(&s3c2412_i2s_dai); -} -module_init(s3c2412_i2s_init); - -static void __exit s3c2412_i2s_exit(void) -{ - snd_soc_unregister_dai(&s3c2412_i2s_dai); -} -module_exit(s3c2412_i2s_exit); - -/* Module information */ -MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("S3C2412 I2S SoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h deleted file mode 100644 index 0b5686b..0000000 --- a/sound/soc/s3c24xx/s3c2412-i2s.h +++ /dev/null @@ -1,29 +0,0 @@ -/* sound/soc/s3c24xx/s3c2412-i2s.c - * - * ALSA Soc Audio Layer - S3C2412 I2S driver - * - * Copyright (c) 2007 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. -*/ - -#ifndef __SND_SOC_S3C24XX_S3C2412_I2S_H -#define __SND_SOC_S3C24XX_S3C2412_I2S_H __FILE__ - -#include "s3c-i2s-v2.h" - -#define S3C2412_DIV_BCLK S3C_I2SV2_DIV_BCLK -#define S3C2412_DIV_RCLK S3C_I2SV2_DIV_RCLK -#define S3C2412_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER - -#define S3C2412_CLKSRC_PCLK S3C_I2SV2_CLKSRC_PCLK -#define S3C2412_CLKSRC_I2SCLK S3C_I2SV2_CLKSRC_AUDIOBUS - -extern struct snd_soc_dai s3c2412_i2s_dai; - -#endif /* __SND_SOC_S3C24XX_S3C2412_I2S_H */ diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c deleted file mode 100644 index c3ac890..0000000 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ /dev/null @@ -1,503 +0,0 @@ -/* - * s3c24xx-i2s.c -- ALSA Soc Audio Layer - * - * (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com - * - * Copyright 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/jiffies.h> -#include <linux/io.h> -#include <linux/gpio.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <mach/hardware.h> -#include <mach/regs-gpio.h> -#include <mach/regs-clock.h> - -#include <asm/dma.h> -#include <mach/dma.h> - -#include <plat/regs-iis.h> - -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" - -static struct s3c2410_dma_client s3c24xx_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c24xx_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - -static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = { - .client = &s3c24xx_dma_client_out, - .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, - .dma_size = 2, -}; - -static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = { - .client = &s3c24xx_dma_client_in, - .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, - .dma_size = 2, -}; - -struct s3c24xx_i2s_info { - void __iomem *regs; - struct clk *iis_clk; - u32 iiscon; - u32 iismod; - u32 iisfcon; - u32 iispsr; -}; -static struct s3c24xx_i2s_info s3c24xx_i2s; - -static void s3c24xx_snd_txctrl(int on) -{ - u32 iisfcon; - u32 iiscon; - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); - iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - - pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); - - if (on) { - iisfcon |= S3C2410_IISFCON_TXDMA | S3C2410_IISFCON_TXENABLE; - iiscon |= S3C2410_IISCON_TXDMAEN | S3C2410_IISCON_IISEN; - iiscon &= ~S3C2410_IISCON_TXIDLE; - iismod |= S3C2410_IISMOD_TXMODE; - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - } else { - /* note, we have to disable the FIFOs otherwise bad things - * seem to happen when the DMA stops. According to the - * Samsung supplied kernel, this should allow the DMA - * engine and FIFOs to reset. If this isn't allowed, the - * DMA engine will simply freeze randomly. - */ - - iisfcon &= ~S3C2410_IISFCON_TXENABLE; - iisfcon &= ~S3C2410_IISFCON_TXDMA; - iiscon |= S3C2410_IISCON_TXIDLE; - iiscon &= ~S3C2410_IISCON_TXDMAEN; - iismod &= ~S3C2410_IISMOD_TXMODE; - - writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - } - - pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); -} - -static void s3c24xx_snd_rxctrl(int on) -{ - u32 iisfcon; - u32 iiscon; - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); - iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - - pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); - - if (on) { - iisfcon |= S3C2410_IISFCON_RXDMA | S3C2410_IISFCON_RXENABLE; - iiscon |= S3C2410_IISCON_RXDMAEN | S3C2410_IISCON_IISEN; - iiscon &= ~S3C2410_IISCON_RXIDLE; - iismod |= S3C2410_IISMOD_RXMODE; - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - } else { - /* note, we have to disable the FIFOs otherwise bad things - * seem to happen when the DMA stops. According to the - * Samsung supplied kernel, this should allow the DMA - * engine and FIFOs to reset. If this isn't allowed, the - * DMA engine will simply freeze randomly. - */ - - iisfcon &= ~S3C2410_IISFCON_RXENABLE; - iisfcon &= ~S3C2410_IISFCON_RXDMA; - iiscon |= S3C2410_IISCON_RXIDLE; - iiscon &= ~S3C2410_IISCON_RXDMAEN; - iismod &= ~S3C2410_IISMOD_RXMODE; - - writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - } - - pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); -} - -/* - * Wait for the LR signal to allow synchronisation to the L/R clock - * from the codec. May only be needed for slave mode. - */ -static int s3c24xx_snd_lrsync(void) -{ - u32 iiscon; - int timeout = 50; /* 5ms */ - - pr_debug("Entered %s\n", __func__); - - while (1) { - iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - if (iiscon & S3C2410_IISCON_LRINDEX) - break; - - if (!timeout--) - return -ETIMEDOUT; - udelay(100); - } - - return 0; -} - -/* - * Check whether CPU is the master or slave - */ -static inline int s3c24xx_snd_is_clkmaster(void) -{ - pr_debug("Entered %s\n", __func__); - - return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1; -} - -/* - * Set S3C24xx I2S DAI format - */ -static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params r: IISMOD: %x \n", iismod); - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - iismod |= S3C2410_IISMOD_SLAVE; - break; - case SND_SOC_DAIFMT_CBS_CFS: - iismod &= ~S3C2410_IISMOD_SLAVE; - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_LEFT_J: - iismod |= S3C2410_IISMOD_MSB; - break; - case SND_SOC_DAIFMT_I2S: - iismod &= ~S3C2410_IISMOD_MSB; - break; - default: - return -EINVAL; - } - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params w: IISMOD: %x \n", iismod); - return 0; -} - -static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data; - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c24xx_i2s_pcm_stereo_out; - else - dma_data = &s3c24xx_i2s_pcm_stereo_in; - - snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_data); - - /* Working copies of register */ - iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params r: IISMOD: %x\n", iismod); - - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - iismod &= ~S3C2410_IISMOD_16BIT; - dma_data->dma_size = 1; - break; - case SNDRV_PCM_FORMAT_S16_LE: - iismod |= S3C2410_IISMOD_16BIT; - dma_data->dma_size = 2; - break; - default: - return -EINVAL; - } - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params w: IISMOD: %x\n", iismod); - return 0; -} - -static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - int ret = 0; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); - - pr_debug("Entered %s\n", __func__); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!s3c24xx_snd_is_clkmaster()) { - ret = s3c24xx_snd_lrsync(); - if (ret) - goto exit_err; - } - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - s3c24xx_snd_rxctrl(1); - else - s3c24xx_snd_txctrl(1); - - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - s3c24xx_snd_rxctrl(0); - else - s3c24xx_snd_txctrl(0); - break; - default: - ret = -EINVAL; - break; - } - -exit_err: - return ret; -} - -/* - * Set S3C24xx Clock source - */ -static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - - pr_debug("Entered %s\n", __func__); - - iismod &= ~S3C2440_IISMOD_MPLL; - - switch (clk_id) { - case S3C24XX_CLKSRC_PCLK: - break; - case S3C24XX_CLKSRC_MPLL: - iismod |= S3C2440_IISMOD_MPLL; - break; - default: - return -EINVAL; - } - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - return 0; -} - -/* - * Set S3C24xx Clock dividers - */ -static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - u32 reg; - - pr_debug("Entered %s\n", __func__); - - switch (div_id) { - case S3C24XX_DIV_BCLK: - reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~S3C2410_IISMOD_FS_MASK; - writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); - break; - case S3C24XX_DIV_MCLK: - reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~(S3C2410_IISMOD_384FS); - writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); - break; - case S3C24XX_DIV_PRESCALER: - writel(div, s3c24xx_i2s.regs + S3C2410_IISPSR); - reg = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - writel(reg | S3C2410_IISCON_PSCEN, s3c24xx_i2s.regs + S3C2410_IISCON); - break; - default: - return -EINVAL; - } - - return 0; -} - -/* - * To avoid duplicating clock code, allow machine driver to - * get the clockrate from here. - */ -u32 s3c24xx_i2s_get_clockrate(void) -{ - return clk_get_rate(s3c24xx_i2s.iis_clk); -} -EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); - -static int s3c24xx_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - pr_debug("Entered %s\n", __func__); - - s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); - if (s3c24xx_i2s.regs == NULL) - return -ENXIO; - - s3c24xx_i2s.iis_clk = clk_get(&pdev->dev, "iis"); - if (s3c24xx_i2s.iis_clk == NULL) { - pr_err("failed to get iis_clock\n"); - iounmap(s3c24xx_i2s.regs); - return -ENODEV; - } - clk_enable(s3c24xx_i2s.iis_clk); - - /* Configure the I2S pins in correct mode */ - s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); - s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); - s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); - - writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON); - - s3c24xx_snd_txctrl(0); - s3c24xx_snd_rxctrl(0); - - return 0; -} - -#ifdef CONFIG_PM -static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai) -{ - pr_debug("Entered %s\n", __func__); - - s3c24xx_i2s.iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - s3c24xx_i2s.iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - s3c24xx_i2s.iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); - s3c24xx_i2s.iispsr = readl(s3c24xx_i2s.regs + S3C2410_IISPSR); - - clk_disable(s3c24xx_i2s.iis_clk); - - return 0; -} - -static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai) -{ - pr_debug("Entered %s\n", __func__); - clk_enable(s3c24xx_i2s.iis_clk); - - writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - writel(s3c24xx_i2s.iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - writel(s3c24xx_i2s.iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(s3c24xx_i2s.iispsr, s3c24xx_i2s.regs + S3C2410_IISPSR); - - return 0; -} -#else -#define s3c24xx_i2s_suspend NULL -#define s3c24xx_i2s_resume NULL -#endif - - -#define S3C24XX_I2S_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - -static struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = { - .trigger = s3c24xx_i2s_trigger, - .hw_params = s3c24xx_i2s_hw_params, - .set_fmt = s3c24xx_i2s_set_fmt, - .set_clkdiv = s3c24xx_i2s_set_clkdiv, - .set_sysclk = s3c24xx_i2s_set_sysclk, -}; - -struct snd_soc_dai s3c24xx_i2s_dai = { - .name = "s3c24xx-i2s", - .id = 0, - .probe = s3c24xx_i2s_probe, - .suspend = s3c24xx_i2s_suspend, - .resume = s3c24xx_i2s_resume, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C24XX_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C24XX_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &s3c24xx_i2s_dai_ops, -}; -EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai); - -static int __init s3c24xx_i2s_init(void) -{ - return snd_soc_register_dai(&s3c24xx_i2s_dai); -} -module_init(s3c24xx_i2s_init); - -static void __exit s3c24xx_i2s_exit(void) -{ - snd_soc_unregister_dai(&s3c24xx_i2s_dai); -} -module_exit(s3c24xx_i2s_exit); - -/* Module information */ -MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("s3c24xx I2S SoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.h b/sound/soc/s3c24xx/s3c24xx-i2s.h deleted file mode 100644 index 726d91c..0000000 --- a/sound/soc/s3c24xx/s3c24xx-i2s.h +++ /dev/null @@ -1,37 +0,0 @@ -/* - * s3c24xx-i2s.c -- ALSA Soc Audio Layer - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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 - * 10th Nov 2006 Initial version. - */ - -#ifndef S3C24XXI2S_H_ -#define S3C24XXI2S_H_ - -/* clock sources */ -#define S3C24XX_CLKSRC_PCLK 0 -#define S3C24XX_CLKSRC_MPLL 1 - -/* Clock dividers */ -#define S3C24XX_DIV_MCLK 0 -#define S3C24XX_DIV_BCLK 1 -#define S3C24XX_DIV_PRESCALER 2 - -/* prescaler */ -#define S3C24XX_PRESCALE(a,b) \ - (((a - 1) << S3C2410_IISPSR_INTSHIFT) | ((b - 1) << S3C2410_IISPSR_EXTSHFIT)) - -u32 s3c24xx_i2s_get_clockrate(void); - -extern struct snd_soc_dai s3c24xx_i2s_dai; - -#endif /*S3C24XXI2S_H_*/ diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c deleted file mode 100644 index 4984754..0000000 --- a/sound/soc/s3c24xx/s3c24xx_simtec.c +++ /dev/null @@ -1,394 +0,0 @@ -/* sound/soc/s3c24xx/s3c24xx_simtec.c - * - * Copyright 2009 Simtec Electronics - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. -*/ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/platform_device.h> -#include <linux/gpio.h> -#include <linux/clk.h> -#include <linux/i2c.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <plat/audio-simtec.h> - -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" -#include "s3c24xx_simtec.h" - -static struct s3c24xx_audio_simtec_pdata *pdata; -static struct clk *xtal_clk; - -static int spk_gain; -static int spk_unmute; - -/** - * speaker_gain_get - read the speaker gain setting. - * @kcontrol: The control for the speaker gain. - * @ucontrol: The value that needs to be updated. - * - * Read the value for the AMP gain control. - */ -static int speaker_gain_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = spk_gain; - return 0; -} - -/** - * speaker_gain_set - set the value of the speaker amp gain - * @value: The value to write. - */ -static void speaker_gain_set(int value) -{ - gpio_set_value_cansleep(pdata->amp_gain[0], value & 1); - gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1); -} - -/** - * speaker_gain_put - set the speaker gain setting. - * @kcontrol: The control for the speaker gain. - * @ucontrol: The value that needs to be set. - * - * Set the value of the speaker gain from the specified - * @ucontrol setting. - * - * Note, if the speaker amp is muted, then we do not set a gain value - * as at-least one of the ICs that is fitted will try and power up even - * if the main control is set to off. - */ -static int speaker_gain_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int value = ucontrol->value.integer.value[0]; - - spk_gain = value; - - if (!spk_unmute) - speaker_gain_set(value); - - return 0; -} - -static const struct snd_kcontrol_new amp_gain_controls[] = { - SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0, - speaker_gain_get, speaker_gain_put), -}; - -/** - * spk_unmute_state - set the unmute state of the speaker - * @to: zero to unmute, non-zero to ununmute. - */ -static void spk_unmute_state(int to) -{ - pr_debug("%s: to=%d\n", __func__, to); - - spk_unmute = to; - gpio_set_value(pdata->amp_gpio, to); - - /* if we're umuting, also re-set the gain */ - if (to && pdata->amp_gain[0] > 0) - speaker_gain_set(spk_gain); -} - -/** - * speaker_unmute_get - read the speaker unmute setting. - * @kcontrol: The control for the speaker gain. - * @ucontrol: The value that needs to be updated. - * - * Read the value for the AMP gain control. - */ -static int speaker_unmute_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = spk_unmute; - return 0; -} - -/** - * speaker_unmute_put - set the speaker unmute setting. - * @kcontrol: The control for the speaker gain. - * @ucontrol: The value that needs to be set. - * - * Set the value of the speaker gain from the specified - * @ucontrol setting. - */ -static int speaker_unmute_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - spk_unmute_state(ucontrol->value.integer.value[0]); - return 0; -} - -/* This is added as a manual control as the speaker amps create clicks - * when their power state is changed, which are far more noticeable than - * anything produced by the CODEC itself. - */ -static const struct snd_kcontrol_new amp_unmute_controls[] = { - SOC_SINGLE_EXT("Speaker Switch", 0, 0, 1, 0, - speaker_unmute_get, speaker_unmute_put), -}; - -void simtec_audio_init(struct snd_soc_codec *codec) -{ - if (pdata->amp_gpio > 0) { - pr_debug("%s: adding amp routes\n", __func__); - - snd_soc_add_controls(codec, amp_unmute_controls, - ARRAY_SIZE(amp_unmute_controls)); - } - - if (pdata->amp_gain[0] > 0) { - pr_debug("%s: adding amp controls\n", __func__); - snd_soc_add_controls(codec, amp_gain_controls, - ARRAY_SIZE(amp_gain_controls)); - } -} -EXPORT_SYMBOL_GPL(simtec_audio_init); - -#define CODEC_CLOCK 12000000 - -/** - * simtec_hw_params - update hardware parameters - * @substream: The audio substream instance. - * @params: The parameters requested. - * - * Update the codec data routing and configuration settings - * from the supplied data. - */ -static int simtec_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; - - /* Set the CODEC as the bus clock master, I2S */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - pr_err("%s: failed set cpu dai format\n", __func__); - return ret; - } - - /* Set the CODEC as the bus clock master */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - pr_err("%s: failed set codec dai format\n", __func__); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, - CODEC_CLOCK, SND_SOC_CLOCK_IN); - if (ret) { - pr_err( "%s: failed setting codec sysclk\n", __func__); - return ret; - } - - if (pdata->use_mpllin) { - ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL, - 0, SND_SOC_CLOCK_OUT); - - if (ret) { - pr_err("%s: failed to set MPLLin as clksrc\n", - __func__); - return ret; - } - } - - if (pdata->output_cdclk) { - int cdclk_scale; - - cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK; - cdclk_scale--; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - cdclk_scale); - } - - return 0; -} - -static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd) -{ - /* call any board supplied startup code, this currently only - * covers the bast/vr1000 which have a CPLD in the way of the - * LRCLK */ - if (pd->startup) - pd->startup(); - - return 0; -} - -static struct snd_soc_ops simtec_snd_ops = { - .hw_params = simtec_hw_params, -}; - -/** - * attach_gpio_amp - get and configure the necessary gpios - * @dev: The device we're probing. - * @pd: The platform data supplied by the board. - * - * If there is a GPIO based amplifier attached to the board, claim - * the necessary GPIO lines for it, and set default values. - */ -static int attach_gpio_amp(struct device *dev, - struct s3c24xx_audio_simtec_pdata *pd) -{ - int ret; - - /* attach gpio amp gain (if any) */ - if (pdata->amp_gain[0] > 0) { - ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain0"); - if (ret) { - dev_err(dev, "cannot get amp gpio gain0\n"); - return ret; - } - - ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain1"); - if (ret) { - dev_err(dev, "cannot get amp gpio gain1\n"); - gpio_free(pdata->amp_gain[0]); - return ret; - } - - gpio_direction_output(pd->amp_gain[0], 0); - gpio_direction_output(pd->amp_gain[1], 0); - } - - /* note, currently we assume GPA0 isn't valid amp */ - if (pdata->amp_gpio > 0) { - ret = gpio_request(pd->amp_gpio, "gpio-amp"); - if (ret) { - dev_err(dev, "cannot get amp gpio %d (%d)\n", - pd->amp_gpio, ret); - goto err_amp; - } - - /* set the amp off at startup */ - spk_unmute_state(0); - } - - return 0; - -err_amp: - if (pd->amp_gain[0] > 0) { - gpio_free(pd->amp_gain[0]); - gpio_free(pd->amp_gain[1]); - } - - return ret; -} - -static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) -{ - if (pd->amp_gain[0] > 0) { - gpio_free(pd->amp_gain[0]); - gpio_free(pd->amp_gain[1]); - } - - if (pd->amp_gpio > 0) - gpio_free(pd->amp_gpio); -} - -#ifdef CONFIG_PM -int simtec_audio_resume(struct device *dev) -{ - simtec_call_startup(pdata); - return 0; -} - -const struct dev_pm_ops simtec_audio_pmops = { - .resume = simtec_audio_resume, -}; -EXPORT_SYMBOL_GPL(simtec_audio_pmops); -#endif - -int __devinit simtec_audio_core_probe(struct platform_device *pdev, - struct snd_soc_device *socdev) -{ - struct platform_device *snd_dev; - int ret; - - socdev->card->dai_link->ops = &simtec_snd_ops; - - pdata = pdev->dev.platform_data; - if (!pdata) { - dev_err(&pdev->dev, "no platform data supplied\n"); - return -EINVAL; - } - - simtec_call_startup(pdata); - - xtal_clk = clk_get(&pdev->dev, "xtal"); - if (IS_ERR(xtal_clk)) { - dev_err(&pdev->dev, "could not get clkout0\n"); - return -EINVAL; - } - - dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk)); - - ret = attach_gpio_amp(&pdev->dev, pdata); - if (ret) - goto err_clk; - - snd_dev = platform_device_alloc("soc-audio", -1); - if (!snd_dev) { - dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n"); - ret = -ENOMEM; - goto err_gpio; - } - - platform_set_drvdata(snd_dev, socdev); - socdev->dev = &snd_dev->dev; - - ret = platform_device_add(snd_dev); - if (ret) { - dev_err(&pdev->dev, "failed to add soc-audio dev\n"); - goto err_pdev; - } - - platform_set_drvdata(pdev, snd_dev); - return 0; - -err_pdev: - platform_device_put(snd_dev); - -err_gpio: - detach_gpio_amp(pdata); - -err_clk: - clk_put(xtal_clk); - return ret; -} -EXPORT_SYMBOL_GPL(simtec_audio_core_probe); - -int __devexit simtec_audio_remove(struct platform_device *pdev) -{ - struct platform_device *snd_dev = platform_get_drvdata(pdev); - - platform_device_unregister(snd_dev); - - detach_gpio_amp(pdata); - clk_put(xtal_clk); - return 0; -} -EXPORT_SYMBOL_GPL(simtec_audio_remove); - -MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("ALSA SoC Simtec Audio common support"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.h b/sound/soc/s3c24xx/s3c24xx_simtec.h deleted file mode 100644 index e18faee..0000000 --- a/sound/soc/s3c24xx/s3c24xx_simtec.h +++ /dev/null @@ -1,22 +0,0 @@ -/* sound/soc/s3c24xx/s3c24xx_simtec.h - * - * Copyright 2009 Simtec Electronics - * - * 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. -*/ - -extern void simtec_audio_init(struct snd_soc_codec *codec); - -extern int simtec_audio_core_probe(struct platform_device *pdev, - struct snd_soc_device *socdev); - -extern int simtec_audio_remove(struct platform_device *pdev); - -#ifdef CONFIG_PM -extern const struct dev_pm_ops simtec_audio_pmops; -#define simtec_audio_pm &simtec_audio_pmops -#else -#define simtec_audio_pm NULL -#endif diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c deleted file mode 100644 index bdf8951..0000000 --- a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c +++ /dev/null @@ -1,153 +0,0 @@ -/* sound/soc/s3c24xx/s3c24xx_simtec_hermes.c - * - * Copyright 2009 Simtec Electronics - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. -*/ - -#include <linux/module.h> -#include <linux/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 <plat/audio-simtec.h> - -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" -#include "s3c24xx_simtec.h" - -#include "../codecs/tlv320aic3x.h" - -static const struct snd_soc_dapm_widget dapm_widgets[] = { - SND_SOC_DAPM_LINE("GSM Out", NULL), - SND_SOC_DAPM_LINE("GSM In", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), - SND_SOC_DAPM_LINE("Line Out", NULL), - SND_SOC_DAPM_LINE("ZV", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), - SND_SOC_DAPM_HP("Headphone Jack", NULL), -}; - -static const struct snd_soc_dapm_route base_map[] = { - /* Headphone connected to HP{L,R}OUT and HP{L,R}COM */ - - { "Headphone Jack", NULL, "HPLOUT" }, - { "Headphone Jack", NULL, "HPLCOM" }, - { "Headphone Jack", NULL, "HPROUT" }, - { "Headphone Jack", NULL, "HPRCOM" }, - - /* ZV connected to Line1 */ - - { "LINE1L", NULL, "ZV" }, - { "LINE1R", NULL, "ZV" }, - - /* Line In connected to Line2 */ - - { "LINE2L", NULL, "Line In" }, - { "LINE2R", NULL, "Line In" }, - - /* Microphone connected to MIC3R and MIC_BIAS */ - - { "MIC3L", NULL, "Mic Jack" }, - - /* GSM connected to MONO_LOUT and MIC3L (in) */ - - { "GSM Out", NULL, "MONO_LOUT" }, - { "MIC3L", NULL, "GSM In" }, - - /* Speaker is connected to LINEOUT{LN,LP,RN,RP}, however we are - * not using the DAPM to power it up and down as there it makes - * a click when powering up. */ -}; - -/** - * simtec_hermes_init - initialise and add controls - * @codec; The codec instance to attach to. - * - * Attach our controls and configure the necessary codec - * mappings for our sound card instance. -*/ -static int simtec_hermes_init(struct snd_soc_codec *codec) -{ - snd_soc_dapm_new_controls(codec, dapm_widgets, - ARRAY_SIZE(dapm_widgets)); - - snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); - - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - snd_soc_dapm_enable_pin(codec, "Line In"); - snd_soc_dapm_enable_pin(codec, "Line Out"); - snd_soc_dapm_enable_pin(codec, "Mic Jack"); - - simtec_audio_init(codec); - snd_soc_dapm_sync(codec); - - return 0; -} - -static struct aic3x_setup_data codec_setup = { -}; - -static struct snd_soc_dai_link simtec_dai_aic33 = { - .name = "tlv320aic33", - .stream_name = "TLV320AIC33", - .cpu_dai = &s3c24xx_i2s_dai, - .codec_dai = &aic3x_dai, - .init = simtec_hermes_init, -}; - -/* simtec audio machine driver */ -static struct snd_soc_card snd_soc_machine_simtec_aic33 = { - .name = "Simtec-Hermes", - .platform = &s3c24xx_soc_platform, - .dai_link = &simtec_dai_aic33, - .num_links = 1, -}; - -/* simtec audio subsystem */ -static struct snd_soc_device simtec_snd_devdata_aic33 = { - .card = &snd_soc_machine_simtec_aic33, - .codec_dev = &soc_codec_dev_aic3x, - .codec_data = &codec_setup, -}; - -static int __devinit simtec_audio_hermes_probe(struct platform_device *pd) -{ - dev_info(&pd->dev, "probing....\n"); - return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic33); -} - -static struct platform_driver simtec_audio_hermes_platdrv = { - .driver = { - .owner = THIS_MODULE, - .name = "s3c24xx-simtec-hermes-snd", - .pm = simtec_audio_pm, - }, - .probe = simtec_audio_hermes_probe, - .remove = __devexit_p(simtec_audio_remove), -}; - -MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd"); - -static int __init simtec_hermes_modinit(void) -{ - return platform_driver_register(&simtec_audio_hermes_platdrv); -} - -static void __exit simtec_hermes_modexit(void) -{ - platform_driver_unregister(&simtec_audio_hermes_platdrv); -} - -module_init(simtec_hermes_modinit); -module_exit(simtec_hermes_modexit); - -MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c deleted file mode 100644 index 185c0ac..0000000 --- a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c +++ /dev/null @@ -1,137 +0,0 @@ -/* sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c - * - * Copyright 2009 Simtec Electronics - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. -*/ - -#include <linux/module.h> -#include <linux/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 <plat/audio-simtec.h> - -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" -#include "s3c24xx_simtec.h" - -#include "../codecs/tlv320aic23.h" - -/* supported machines: - * - * Machine Connections AMP - * ------- ----------- --- - * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired) - * VR1000 HPOUT, LIN None - * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) - * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) - * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R) - */ - -static const struct snd_soc_dapm_widget dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), - SND_SOC_DAPM_LINE("Line Out", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), -}; - -static const struct snd_soc_dapm_route base_map[] = { - { "Headphone Jack", NULL, "LHPOUT"}, - { "Headphone Jack", NULL, "RHPOUT"}, - - { "Line Out", NULL, "LOUT" }, - { "Line Out", NULL, "ROUT" }, - - { "LLINEIN", NULL, "Line In"}, - { "RLINEIN", NULL, "Line In"}, - - { "MICIN", NULL, "Mic Jack"}, -}; - -/** - * simtec_tlv320aic23_init - initialise and add controls - * @codec; The codec instance to attach to. - * - * Attach our controls and configure the necessary codec - * mappings for our sound card instance. -*/ -static int simtec_tlv320aic23_init(struct snd_soc_codec *codec) -{ - snd_soc_dapm_new_controls(codec, dapm_widgets, - ARRAY_SIZE(dapm_widgets)); - - snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); - - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - snd_soc_dapm_enable_pin(codec, "Line In"); - snd_soc_dapm_enable_pin(codec, "Line Out"); - snd_soc_dapm_enable_pin(codec, "Mic Jack"); - - simtec_audio_init(codec); - snd_soc_dapm_sync(codec); - - return 0; -} - -static struct snd_soc_dai_link simtec_dai_aic23 = { - .name = "tlv320aic23", - .stream_name = "TLV320AIC23", - .cpu_dai = &s3c24xx_i2s_dai, - .codec_dai = &tlv320aic23_dai, - .init = simtec_tlv320aic23_init, -}; - -/* simtec audio machine driver */ -static struct snd_soc_card snd_soc_machine_simtec_aic23 = { - .name = "Simtec", - .platform = &s3c24xx_soc_platform, - .dai_link = &simtec_dai_aic23, - .num_links = 1, -}; - -/* simtec audio subsystem */ -static struct snd_soc_device simtec_snd_devdata_aic23 = { - .card = &snd_soc_machine_simtec_aic23, - .codec_dev = &soc_codec_dev_tlv320aic23, -}; - -static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd) -{ - return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic23); -} - -static struct platform_driver simtec_audio_tlv320aic23_platdrv = { - .driver = { - .owner = THIS_MODULE, - .name = "s3c24xx-simtec-tlv320aic23", - .pm = simtec_audio_pm, - }, - .probe = simtec_audio_tlv320aic23_probe, - .remove = __devexit_p(simtec_audio_remove), -}; - -MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); - -static int __init simtec_tlv320aic23_modinit(void) -{ - return platform_driver_register(&simtec_audio_tlv320aic23_platdrv); -} - -static void __exit simtec_tlv320aic23_modexit(void) -{ - platform_driver_unregister(&simtec_audio_tlv320aic23_platdrv); -} - -module_init(simtec_tlv320aic23_modinit); -module_exit(simtec_tlv320aic23_modexit); - -MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c deleted file mode 100644 index 052d596..0000000 --- a/sound/soc/s3c24xx/s3c24xx_uda134x.c +++ /dev/null @@ -1,373 +0,0 @@ -/* - * Modifications by Christian Pellegrin <chripell(a)evolware.org> - * - * s3c24xx_uda134x.c -- S3C24XX_UDA134X ALSA SoC Audio board driver - * - * Copyright 2007 Dension Audio Systems Ltd. - * Author: Zoltan Devai - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/clk.h> -#include <linux/mutex.h> -#include <linux/gpio.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> -#include <sound/s3c24xx_uda134x.h> -#include <sound/uda134x.h> - -#include <plat/regs-iis.h> - -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" -#include "../codecs/uda134x.h" - - -/* #define ENFORCE_RATES 1 */ -/* - Unfortunately the S3C24XX in master mode has a limited capacity of - generating the clock for the codec. If you define this only rates - that are really available will be enforced. But be careful, most - user level application just want the usual sampling frequencies (8, - 11.025, 22.050, 44.1 kHz) and anyway resampling is a costly - operation for embedded systems. So if you aren't very lucky or your - hardware engineer wasn't very forward-looking it's better to leave - this undefined. If you do so an approximate value for the requested - sampling rate in the range -/+ 5% will be chosen. If this in not - possible an error will be returned. -*/ - -static struct clk *xtal; -static struct clk *pclk; -/* this is need because we don't have a place where to keep the - * pointers to the clocks in each substream. We get the clocks only - * when we are actually using them so we don't block stuff like - * frequency change or oscillator power-off */ -static int clk_users; -static DEFINE_MUTEX(clk_lock); - -static unsigned int rates[33 * 2]; -#ifdef ENFORCE_RATES -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; -#endif - -static struct platform_device *s3c24xx_uda134x_snd_device; - -static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) -{ - int ret = 0; -#ifdef ENFORCE_RATES - struct snd_pcm_runtime *runtime = substream->runtime; -#endif - - mutex_lock(&clk_lock); - pr_debug("%s %d\n", __func__, clk_users); - if (clk_users == 0) { - xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal"); - if (!xtal) { - printk(KERN_ERR "%s cannot get xtal\n", __func__); - ret = -EBUSY; - } else { - pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, - "pclk"); - if (!pclk) { - printk(KERN_ERR "%s cannot get pclk\n", - __func__); - clk_put(xtal); - ret = -EBUSY; - } - } - if (!ret) { - int i, j; - - for (i = 0; i < 2; i++) { - int fs = i ? 256 : 384; - - rates[i*33] = clk_get_rate(xtal) / fs; - for (j = 1; j < 33; j++) - rates[i*33 + j] = clk_get_rate(pclk) / - (j * fs); - } - } - } - clk_users += 1; - mutex_unlock(&clk_lock); - if (!ret) { -#ifdef ENFORCE_RATES - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &hw_constraints_rates); - if (ret < 0) - printk(KERN_ERR "%s cannot set constraints\n", - __func__); -#endif - } - return ret; -} - -static void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream) -{ - mutex_lock(&clk_lock); - pr_debug("%s %d\n", __func__, clk_users); - clk_users -= 1; - if (clk_users == 0) { - clk_put(xtal); - xtal = NULL; - clk_put(pclk); - pclk = NULL; - } - mutex_unlock(&clk_lock); -} - -static int s3c24xx_uda134x_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; - unsigned int clk = 0; - int ret = 0; - int clk_source, fs_mode; - unsigned long rate = params_rate(params); - long err, cerr; - unsigned int div; - int i, bi; - - err = 999999; - bi = 0; - for (i = 0; i < 2*33; i++) { - cerr = rates[i] - rate; - if (cerr < 0) - cerr = -cerr; - if (cerr < err) { - err = cerr; - bi = i; - } - } - if (bi / 33 == 1) - fs_mode = S3C2410_IISMOD_256FS; - else - fs_mode = S3C2410_IISMOD_384FS; - if (bi % 33 == 0) { - clk_source = S3C24XX_CLKSRC_MPLL; - div = 1; - } else { - clk_source = S3C24XX_CLKSRC_PCLK; - div = bi % 33; - } - pr_debug("%s desired rate %lu, %d\n", __func__, rate, bi); - - clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate; - pr_debug("%s will use: %s %s %d sysclk %d err %ld\n", __func__, - fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS", - clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK", - div, clk, err); - - if ((err * 100 / rate) > 5) { - printk(KERN_ERR "S3C24XX_UDA134X: effective frequency " - "too different from desired (%ld%%)\n", - err * 100 / rate); - return -EINVAL; - } - - 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; - - 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; - - ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source , clk, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, fs_mode); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, - S3C2410_IISMOD_32FS); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - S3C24XX_PRESCALE(div, div)); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, - SND_SOC_CLOCK_OUT); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops s3c24xx_uda134x_ops = { - .startup = s3c24xx_uda134x_startup, - .shutdown = s3c24xx_uda134x_shutdown, - .hw_params = s3c24xx_uda134x_hw_params, -}; - -static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { - .name = "UDA134X", - .stream_name = "UDA134X", - .codec_dai = &uda134x_dai, - .cpu_dai = &s3c24xx_i2s_dai, - .ops = &s3c24xx_uda134x_ops, -}; - -static struct snd_soc_card snd_soc_s3c24xx_uda134x = { - .name = "S3C24XX_UDA134X", - .platform = &s3c24xx_soc_platform, - .dai_link = &s3c24xx_uda134x_dai_link, - .num_links = 1, -}; - -static struct s3c24xx_uda134x_platform_data *s3c24xx_uda134x_l3_pins; - -static void setdat(int v) -{ - gpio_set_value(s3c24xx_uda134x_l3_pins->l3_data, v > 0); -} - -static void setclk(int v) -{ - gpio_set_value(s3c24xx_uda134x_l3_pins->l3_clk, v > 0); -} - -static void setmode(int v) -{ - gpio_set_value(s3c24xx_uda134x_l3_pins->l3_mode, v > 0); -} - -static struct uda134x_platform_data s3c24xx_uda134x = { - .l3 = { - .setdat = setdat, - .setclk = setclk, - .setmode = setmode, - .data_hold = 1, - .data_setup = 1, - .clock_high = 1, - .mode_hold = 1, - .mode = 1, - .mode_setup = 1, - }, -}; - -static struct snd_soc_device s3c24xx_uda134x_snd_devdata = { - .card = &snd_soc_s3c24xx_uda134x, - .codec_dev = &soc_codec_dev_uda134x, - .codec_data = &s3c24xx_uda134x, -}; - -static int s3c24xx_uda134x_setup_pin(int pin, char *fun) -{ - if (gpio_request(pin, "s3c24xx_uda134x") < 0) { - printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " - "l3 %s pin already in use", fun); - return -EBUSY; - } - gpio_direction_output(pin, 0); - return 0; -} - -static int s3c24xx_uda134x_probe(struct platform_device *pdev) -{ - int ret; - - printk(KERN_INFO "S3C24XX_UDA134X SoC Audio driver\n"); - - s3c24xx_uda134x_l3_pins = pdev->dev.platform_data; - if (s3c24xx_uda134x_l3_pins == NULL) { - printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " - "unable to find platform data\n"); - return -ENODEV; - } - s3c24xx_uda134x.power = s3c24xx_uda134x_l3_pins->power; - s3c24xx_uda134x.model = s3c24xx_uda134x_l3_pins->model; - - if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_data, - "data") < 0) - return -EBUSY; - if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_clk, - "clk") < 0) { - gpio_free(s3c24xx_uda134x_l3_pins->l3_data); - return -EBUSY; - } - if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_mode, - "mode") < 0) { - gpio_free(s3c24xx_uda134x_l3_pins->l3_data); - gpio_free(s3c24xx_uda134x_l3_pins->l3_clk); - return -EBUSY; - } - - s3c24xx_uda134x_snd_device = platform_device_alloc("soc-audio", -1); - if (!s3c24xx_uda134x_snd_device) { - printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " - "Unable to register\n"); - return -ENOMEM; - } - - platform_set_drvdata(s3c24xx_uda134x_snd_device, - &s3c24xx_uda134x_snd_devdata); - s3c24xx_uda134x_snd_devdata.dev = &s3c24xx_uda134x_snd_device->dev; - ret = platform_device_add(s3c24xx_uda134x_snd_device); - if (ret) { - printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n"); - platform_device_put(s3c24xx_uda134x_snd_device); - } - - return ret; -} - -static int s3c24xx_uda134x_remove(struct platform_device *pdev) -{ - platform_device_unregister(s3c24xx_uda134x_snd_device); - gpio_free(s3c24xx_uda134x_l3_pins->l3_data); - gpio_free(s3c24xx_uda134x_l3_pins->l3_clk); - gpio_free(s3c24xx_uda134x_l3_pins->l3_mode); - return 0; -} - -static struct platform_driver s3c24xx_uda134x_driver = { - .probe = s3c24xx_uda134x_probe, - .remove = s3c24xx_uda134x_remove, - .driver = { - .name = "s3c24xx_uda134x", - .owner = THIS_MODULE, - }, -}; - -static int __init s3c24xx_uda134x_init(void) -{ - return platform_driver_register(&s3c24xx_uda134x_driver); -} - -static void __exit s3c24xx_uda134x_exit(void) -{ - platform_driver_unregister(&s3c24xx_uda134x_driver); -} - - -module_init(s3c24xx_uda134x_init); -module_exit(s3c24xx_uda134x_exit); - -MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>"); -MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c64xx-i2s-v4.c b/sound/soc/s3c24xx/s3c64xx-i2s-v4.c deleted file mode 100644 index 06db130..0000000 --- a/sound/soc/s3c24xx/s3c64xx-i2s-v4.c +++ /dev/null @@ -1,209 +0,0 @@ -/* sound/soc/s3c24xx/s3c64xx-i2s-v4.c - * - * ALSA SoC Audio Layer - S3C64XX I2Sv4 driver - * Copyright (c) 2010 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.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. - */ - -#include <linux/clk.h> -#include <linux/gpio.h> -#include <linux/io.h> - -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include <mach/gpio-bank-c.h> -#include <mach/gpio-bank-h.h> -#include <plat/gpio-cfg.h> - -#include <mach/map.h> -#include <mach/dma.h> - -#include "s3c-dma.h" -#include "regs-i2s-v2.h" -#include "s3c64xx-i2s.h" - -static struct s3c2410_dma_client s3c64xx_dma_client_out = { - .name = "I2Sv4 PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c64xx_dma_client_in = { - .name = "I2Sv4 PCM Stereo in" -}; - -static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_out; -static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_in; -static struct s3c_i2sv2_info s3c64xx_i2sv4; - -struct snd_soc_dai s3c64xx_i2s_v4_dai; -EXPORT_SYMBOL_GPL(s3c64xx_i2s_v4_dai); - -static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) -{ - return cpu_dai->private_data; -} - -static int s3c64xx_i2sv4_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - /* configure GPIO for i2s port */ - s3c_gpio_cfgpin(S3C64XX_GPC(4), S3C64XX_GPC4_I2S_V40_DO0); - s3c_gpio_cfgpin(S3C64XX_GPC(5), S3C64XX_GPC5_I2S_V40_DO1); - s3c_gpio_cfgpin(S3C64XX_GPC(7), S3C64XX_GPC7_I2S_V40_DO2); - s3c_gpio_cfgpin(S3C64XX_GPH(6), S3C64XX_GPH6_I2S_V40_BCLK); - s3c_gpio_cfgpin(S3C64XX_GPH(7), S3C64XX_GPH7_I2S_V40_CDCLK); - s3c_gpio_cfgpin(S3C64XX_GPH(8), S3C64XX_GPH8_I2S_V40_LRCLK); - s3c_gpio_cfgpin(S3C64XX_GPH(9), S3C64XX_GPH9_I2S_V40_DI); - - return 0; -} - -static int s3c_i2sv4_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - struct s3c_dma_params *dma_data; - u32 iismod; - - dev_dbg(cpu_dai->dev, "Entered %s\n", __func__); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = i2s->dma_playback; - else - dma_data = i2s->dma_capture; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - iismod = readl(i2s->regs + S3C2412_IISMOD); - dev_dbg(cpu_dai->dev, "%s: r: IISMOD: %x\n", __func__, iismod); - - iismod &= ~S3C64XX_IISMOD_BLC_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - iismod |= S3C64XX_IISMOD_BLC_8BIT; - break; - case SNDRV_PCM_FORMAT_S16_LE: - break; - case SNDRV_PCM_FORMAT_S24_LE: - iismod |= S3C64XX_IISMOD_BLC_24BIT; - break; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - dev_dbg(cpu_dai->dev, "%s: w: IISMOD: %x\n", __func__, iismod); - - return 0; -} - -static struct snd_soc_dai_ops s3c64xx_i2sv4_dai_ops = { - .hw_params = s3c_i2sv4_hw_params, -}; - -static __devinit int s3c64xx_i2sv4_dev_probe(struct platform_device *pdev) -{ - struct s3c_i2sv2_info *i2s; - struct snd_soc_dai *dai; - int ret; - - i2s = &s3c64xx_i2sv4; - dai = &s3c64xx_i2s_v4_dai; - - if (dai->dev) { - dev_dbg(dai->dev, "%s: \ - I2Sv4 instance already registered!\n", __func__); - return -EBUSY; - } - - dai->dev = &pdev->dev; - dai->name = "s3c64xx-i2s-v4"; - dai->id = 0; - dai->symmetric_rates = 1; - dai->playback.channels_min = 2; - dai->playback.channels_max = 2; - dai->playback.rates = S3C64XX_I2S_RATES; - dai->playback.formats = S3C64XX_I2S_FMTS; - dai->capture.channels_min = 2; - dai->capture.channels_max = 2; - dai->capture.rates = S3C64XX_I2S_RATES; - dai->capture.formats = S3C64XX_I2S_FMTS; - dai->probe = s3c64xx_i2sv4_probe; - dai->ops = &s3c64xx_i2sv4_dai_ops; - - i2s->feature |= S3C_FEATURE_CDCLKCON; - - i2s->dma_capture = &s3c64xx_i2sv4_pcm_stereo_in; - i2s->dma_playback = &s3c64xx_i2sv4_pcm_stereo_out; - - i2s->dma_capture->channel = DMACH_HSI_I2SV40_RX; - i2s->dma_capture->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISRXD; - i2s->dma_playback->channel = DMACH_HSI_I2SV40_TX; - i2s->dma_playback->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISTXD; - - i2s->dma_capture->client = &s3c64xx_dma_client_in; - i2s->dma_capture->dma_size = 4; - i2s->dma_playback->client = &s3c64xx_dma_client_out; - i2s->dma_playback->dma_size = 4; - - i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); - if (IS_ERR(i2s->iis_cclk)) { - dev_err(&pdev->dev, "failed to get audio-bus\n"); - ret = PTR_ERR(i2s->iis_cclk); - goto err; - } - - clk_enable(i2s->iis_cclk); - - ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); - if (ret) - goto err_clk; - - ret = s3c_i2sv2_register_dai(dai); - if (ret != 0) - goto err_i2sv2; - - return 0; - -err_i2sv2: - /* Not implemented for I2Sv2 core yet */ -err_clk: - clk_put(i2s->iis_cclk); -err: - return ret; -} - -static __devexit int s3c64xx_i2sv4_dev_remove(struct platform_device *pdev) -{ - dev_err(&pdev->dev, "Device removal not yet supported\n"); - return 0; -} - -static struct platform_driver s3c64xx_i2sv4_driver = { - .probe = s3c64xx_i2sv4_dev_probe, - .remove = s3c64xx_i2sv4_dev_remove, - .driver = { - .name = "s3c64xx-iis-v4", - .owner = THIS_MODULE, - }, -}; - -static int __init s3c64xx_i2sv4_init(void) -{ - return platform_driver_register(&s3c64xx_i2sv4_driver); -} -module_init(s3c64xx_i2sv4_init); - -static void __exit s3c64xx_i2sv4_exit(void) -{ - platform_driver_unregister(&s3c64xx_i2sv4_driver); -} -module_exit(s3c64xx_i2sv4_exit); - -/* Module information */ -MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); -MODULE_DESCRIPTION("S3C64XX I2Sv4 SoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c deleted file mode 100644 index 1d85cb8..0000000 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ /dev/null @@ -1,190 +0,0 @@ -/* sound/soc/s3c24xx/s3c64xx-i2s.c - * - * ALSA SoC Audio Layer - S3C64XX I2S driver - * - * Copyright 2008 Openmoko, Inc. - * Copyright 2008 Simtec Electronics - * Ben Dooks <ben(a)simtec.co.uk> - * http://armlinux.simtec.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 - * published by the Free Software Foundation. - */ - -#include <linux/clk.h> -#include <linux/gpio.h> -#include <linux/io.h> - -#include <sound/soc.h> - -#include <mach/gpio-bank-d.h> -#include <mach/gpio-bank-e.h> -#include <plat/gpio-cfg.h> - -#include <mach/map.h> -#include <mach/dma.h> - -#include "s3c-dma.h" -#include "regs-i2s-v2.h" -#include "s3c64xx-i2s.h" - -/* The value should be set to maximum of the total number - * of I2Sv3 controllers that any supported SoC has. - */ -#define MAX_I2SV3 2 - -static struct s3c2410_dma_client s3c64xx_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c64xx_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - -static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_out[MAX_I2SV3]; -static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_in[MAX_I2SV3]; -static struct s3c_i2sv2_info s3c64xx_i2s[MAX_I2SV3]; - -struct snd_soc_dai s3c64xx_i2s_dai[MAX_I2SV3]; -EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); - -static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) -{ - return cpu_dai->private_data; -} - -static int s3c64xx_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - /* configure GPIO for i2s port */ - switch (dai->id) { - case 0: - s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK); - s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK); - s3c_gpio_cfgpin(S3C64XX_GPD(2), S3C64XX_GPD2_I2S0_LRCLK); - s3c_gpio_cfgpin(S3C64XX_GPD(3), S3C64XX_GPD3_I2S0_DI); - s3c_gpio_cfgpin(S3C64XX_GPD(4), S3C64XX_GPD4_I2S0_D0); - break; - case 1: - s3c_gpio_cfgpin(S3C64XX_GPE(0), S3C64XX_GPE0_I2S1_CLK); - s3c_gpio_cfgpin(S3C64XX_GPE(1), S3C64XX_GPE1_I2S1_CDCLK); - s3c_gpio_cfgpin(S3C64XX_GPE(2), S3C64XX_GPE2_I2S1_LRCLK); - s3c_gpio_cfgpin(S3C64XX_GPE(3), S3C64XX_GPE3_I2S1_DI); - s3c_gpio_cfgpin(S3C64XX_GPE(4), S3C64XX_GPE4_I2S1_D0); - } - - return 0; -} - - -static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops; - -static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) -{ - struct s3c_i2sv2_info *i2s; - struct snd_soc_dai *dai; - int ret; - - if (pdev->id >= MAX_I2SV3) { - dev_err(&pdev->dev, "id %d out of range\n", pdev->id); - return -EINVAL; - } - - i2s = &s3c64xx_i2s[pdev->id]; - dai = &s3c64xx_i2s_dai[pdev->id]; - dai->dev = &pdev->dev; - dai->name = "s3c64xx-i2s"; - dai->id = pdev->id; - dai->symmetric_rates = 1; - dai->playback.channels_min = 2; - dai->playback.channels_max = 2; - dai->playback.rates = S3C64XX_I2S_RATES; - dai->playback.formats = S3C64XX_I2S_FMTS; - dai->capture.channels_min = 2; - dai->capture.channels_max = 2; - dai->capture.rates = S3C64XX_I2S_RATES; - dai->capture.formats = S3C64XX_I2S_FMTS; - dai->probe = s3c64xx_i2s_probe; - dai->ops = &s3c64xx_i2s_dai_ops; - - i2s->feature |= S3C_FEATURE_CDCLKCON; - - i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id]; - i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id]; - - if (pdev->id == 0) { - i2s->dma_capture->channel = DMACH_I2S0_IN; - i2s->dma_capture->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISRXD; - i2s->dma_playback->channel = DMACH_I2S0_OUT; - i2s->dma_playback->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISTXD; - } else { - i2s->dma_capture->channel = DMACH_I2S1_IN; - i2s->dma_capture->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISRXD; - i2s->dma_playback->channel = DMACH_I2S1_OUT; - i2s->dma_playback->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISTXD; - } - - i2s->dma_capture->client = &s3c64xx_dma_client_in; - i2s->dma_capture->dma_size = 4; - i2s->dma_playback->client = &s3c64xx_dma_client_out; - i2s->dma_playback->dma_size = 4; - - i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); - if (IS_ERR(i2s->iis_cclk)) { - dev_err(&pdev->dev, "failed to get audio-bus\n"); - ret = PTR_ERR(i2s->iis_cclk); - goto err; - } - - clk_enable(i2s->iis_cclk); - - ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); - if (ret) - goto err_clk; - - ret = s3c_i2sv2_register_dai(dai); - if (ret != 0) - goto err_i2sv2; - - return 0; - -err_i2sv2: - /* Not implemented for I2Sv2 core yet */ -err_clk: - clk_put(i2s->iis_cclk); -err: - return ret; -} - -static __devexit int s3c64xx_iis_dev_remove(struct platform_device *pdev) -{ - dev_err(&pdev->dev, "Device removal not yet supported\n"); - return 0; -} - -static struct platform_driver s3c64xx_iis_driver = { - .probe = s3c64xx_iis_dev_probe, - .remove = s3c64xx_iis_dev_remove, - .driver = { - .name = "s3c64xx-iis", - .owner = THIS_MODULE, - }, -}; - -static int __init s3c64xx_i2s_init(void) -{ - return platform_driver_register(&s3c64xx_iis_driver); -} -module_init(s3c64xx_i2s_init); - -static void __exit s3c64xx_i2s_exit(void) -{ - platform_driver_unregister(&s3c64xx_iis_driver); -} -module_exit(s3c64xx_i2s_exit); - -/* Module information */ -MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("S3C64XX I2S SoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h deleted file mode 100644 index 7a40f43..0000000 --- a/sound/soc/s3c24xx/s3c64xx-i2s.h +++ /dev/null @@ -1,42 +0,0 @@ -/* sound/soc/s3c24xx/s3c64xx-i2s.h - * - * ALSA SoC Audio Layer - S3C64XX I2S driver - * - * Copyright 2008 Openmoko, Inc. - * Copyright 2008 Simtec Electronics - * Ben Dooks <ben(a)simtec.co.uk> - * http://armlinux.simtec.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 - * published by the Free Software Foundation. - */ - -#ifndef __SND_SOC_S3C24XX_S3C64XX_I2S_H -#define __SND_SOC_S3C24XX_S3C64XX_I2S_H __FILE__ - -struct clk; - -#include "s3c-i2s-v2.h" - -#define S3C64XX_DIV_BCLK S3C_I2SV2_DIV_BCLK -#define S3C64XX_DIV_RCLK S3C_I2SV2_DIV_RCLK -#define S3C64XX_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER - -#define S3C64XX_CLKSRC_PCLK S3C_I2SV2_CLKSRC_PCLK -#define S3C64XX_CLKSRC_MUX S3C_I2SV2_CLKSRC_AUDIOBUS -#define S3C64XX_CLKSRC_CDCLK S3C_I2SV2_CLKSRC_CDCLK - -#define S3C64XX_I2S_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - -#define S3C64XX_I2S_FMTS \ - (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE) - -extern struct snd_soc_dai s3c64xx_i2s_dai[]; -extern struct snd_soc_dai s3c64xx_i2s_v4_dai; - -#endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */ diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c deleted file mode 100644 index 3622588..0000000 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ /dev/null @@ -1,81 +0,0 @@ -/* - * smdk2443_wm9710.c -- SoC audio for smdk2443 - * - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include "../codecs/ac97.h" -#include "s3c-dma.h" -#include "s3c-ac97.h" - -static struct snd_soc_card smdk2443; - -static struct snd_soc_dai_link smdk2443_dai[] = { -{ - .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], - .codec_dai = &ac97_dai, -}, -}; - -static struct snd_soc_card smdk2443 = { - .name = "SMDK2443", - .platform = &s3c24xx_soc_platform, - .dai_link = smdk2443_dai, - .num_links = ARRAY_SIZE(smdk2443_dai), -}; - -static struct snd_soc_device smdk2443_snd_ac97_devdata = { - .card = &smdk2443, - .codec_dev = &soc_codec_dev_ac97, -}; - -static struct platform_device *smdk2443_snd_ac97_device; - -static int __init smdk2443_init(void) -{ - int ret; - - smdk2443_snd_ac97_device = platform_device_alloc("soc-audio", -1); - if (!smdk2443_snd_ac97_device) - return -ENOMEM; - - platform_set_drvdata(smdk2443_snd_ac97_device, - &smdk2443_snd_ac97_devdata); - smdk2443_snd_ac97_devdata.dev = &smdk2443_snd_ac97_device->dev; - ret = platform_device_add(smdk2443_snd_ac97_device); - - if (ret) - platform_device_put(smdk2443_snd_ac97_device); - - return ret; -} - -static void __exit smdk2443_exit(void) -{ - platform_device_unregister(smdk2443_snd_ac97_device); -} - -module_init(smdk2443_init); -module_exit(smdk2443_exit); - -/* Module information */ -MODULE_AUTHOR("Graeme Gregory, graeme.gregory(a)wolfsonmicro.com, www.wolfsonmicro.com"); -MODULE_DESCRIPTION("ALSA SoC WM9710 SMDK2443"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c deleted file mode 100644 index 07e8e51..0000000 --- a/sound/soc/s3c24xx/smdk64xx_wm8580.c +++ /dev/null @@ -1,266 +0,0 @@ -/* - * smdk64xx_wm8580.c - * - * Copyright (c) 2009 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.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/platform_device.h> -#include <linux/clk.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include "../codecs/wm8580.h" -#include "s3c-dma.h" -#include "s3c64xx-i2s.h" - -/* SMDK64XX has a 12MHZ crystal attached to WM8580 */ -#define SMDK64XX_WM8580_FREQ 12000000 - -static int smdk64xx_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; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - unsigned int pll_out; - int bfs, rfs, ret; - - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_U8: - case SNDRV_PCM_FORMAT_S8: - bfs = 16; - break; - case SNDRV_PCM_FORMAT_U16_LE: - case SNDRV_PCM_FORMAT_S16_LE: - bfs = 32; - break; - default: - return -EINVAL; - } - - /* The Fvco for WM8580 PLLs must fall within [90,100]MHz. - * This criterion can't be met if we request PLL output - * as {8000x256, 64000x256, 11025x256}Hz. - * As a wayout, we rather change rfs to a minimum value that - * results in (params_rate(params) * rfs), and itself, acceptable - * to both - the CODEC and the CPU. - */ - switch (params_rate(params)) { - case 16000: - case 22050: - case 32000: - case 44100: - case 48000: - case 88200: - case 96000: - rfs = 256; - break; - case 64000: - rfs = 384; - break; - case 8000: - case 11025: - rfs = 512; - break; - default: - return -EINVAL; - } - pll_out = params_rate(params) * rfs; - - /* Set the 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 the AP 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; - - ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_CDCLK, - 0, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* We use PCLK for basic ops in SoC-Slave mode */ - ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, - 0, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* Set WM8580 to drive MCLK from its PLLA */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, - WM8580_CLKSRC_PLLA); - if (ret < 0) - return ret; - - /* Explicitly set WM8580-DAC to source from MCLK */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL, - WM8580_CLKSRC_MCLK); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, - SMDK64XX_WM8580_FREQ, pll_out); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs); - if (ret < 0) - return ret; - - return 0; -} - -/* - * SMDK64XX WM8580 DAI operations. - */ -static struct snd_soc_ops smdk64xx_ops = { - .hw_params = smdk64xx_hw_params, -}; - -/* SMDK64xx Playback widgets */ -static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { - SND_SOC_DAPM_HP("Front-L/R", NULL), - SND_SOC_DAPM_HP("Center/Sub", NULL), - SND_SOC_DAPM_HP("Rear-L/R", NULL), -}; - -/* SMDK64xx Capture widgets */ -static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { - SND_SOC_DAPM_MIC("MicIn", NULL), - SND_SOC_DAPM_LINE("LineIn", NULL), -}; - -/* SMDK-PAIFTX connections */ -static const struct snd_soc_dapm_route audio_map_tx[] = { - /* MicIn feeds AINL */ - {"AINL", NULL, "MicIn"}, - - /* LineIn feeds AINL/R */ - {"AINL", NULL, "LineIn"}, - {"AINR", NULL, "LineIn"}, -}; - -/* SMDK-PAIFRX connections */ -static const struct snd_soc_dapm_route audio_map_rx[] = { - /* Front Left/Right are fed VOUT1L/R */ - {"Front-L/R", NULL, "VOUT1L"}, - {"Front-L/R", NULL, "VOUT1R"}, - - /* Center/Sub are fed VOUT2L/R */ - {"Center/Sub", NULL, "VOUT2L"}, - {"Center/Sub", NULL, "VOUT2R"}, - - /* Rear Left/Right are fed VOUT3L/R */ - {"Rear-L/R", NULL, "VOUT3L"}, - {"Rear-L/R", NULL, "VOUT3R"}, -}; - -static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec) -{ - /* Add smdk64xx specific Capture widgets */ - snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt, - ARRAY_SIZE(wm8580_dapm_widgets_cpt)); - - /* Set up PAIFTX audio path */ - snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); - - /* Enabling the microphone requires the fitting of a 0R - * resistor to connect the line from the microphone jack. - */ - snd_soc_dapm_disable_pin(codec, "MicIn"); - - /* signal a DAPM event */ - snd_soc_dapm_sync(codec); - - return 0; -} - -static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec) -{ - /* Add smdk64xx specific Playback widgets */ - snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk, - ARRAY_SIZE(wm8580_dapm_widgets_pbk)); - - /* Set up PAIFRX audio path */ - snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); - - /* signal a DAPM event */ - snd_soc_dapm_sync(codec); - - return 0; -} - -static struct snd_soc_dai_link smdk64xx_dai[] = { -{ /* Primary Playback i/f */ - .name = "WM8580 PAIF RX", - .stream_name = "Playback", - .cpu_dai = &s3c64xx_i2s_v4_dai, - .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], - .init = smdk64xx_wm8580_init_paifrx, - .ops = &smdk64xx_ops, -}, -{ /* Primary Capture i/f */ - .name = "WM8580 PAIF TX", - .stream_name = "Capture", - .cpu_dai = &s3c64xx_i2s_v4_dai, - .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], - .init = smdk64xx_wm8580_init_paiftx, - .ops = &smdk64xx_ops, -}, -}; - -static struct snd_soc_card smdk64xx = { - .name = "smdk64xx", - .platform = &s3c24xx_soc_platform, - .dai_link = smdk64xx_dai, - .num_links = ARRAY_SIZE(smdk64xx_dai), -}; - -static struct snd_soc_device smdk64xx_snd_devdata = { - .card = &smdk64xx, - .codec_dev = &soc_codec_dev_wm8580, -}; - -static struct platform_device *smdk64xx_snd_device; - -static int __init smdk64xx_audio_init(void) -{ - int ret; - - smdk64xx_snd_device = platform_device_alloc("soc-audio", -1); - if (!smdk64xx_snd_device) - return -ENOMEM; - - platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata); - smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev; - ret = platform_device_add(smdk64xx_snd_device); - - if (ret) - platform_device_put(smdk64xx_snd_device); - - return ret; -} -module_init(smdk64xx_audio_init); - -MODULE_AUTHOR("Jaswinder Singh, jassi.brar(a)samsung.com"); -MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c deleted file mode 100644 index 24fd39f..0000000 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ /dev/null @@ -1,94 +0,0 @@ -/* - * smdk_wm9713.c -- SoC audio for SMDK - * - * Copyright 2010 Samsung Electronics Co. Ltd. - * Author: Jaswinder Singh Brar <jassi.brar(a)samsung.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/device.h> -#include <sound/soc.h> - -#include "../codecs/wm9713.h" -#include "s3c-dma.h" -#include "s3c-ac97.h" - -static struct snd_soc_card smdk; - -/* - * Default CFG switch settings to use this driver: - * - * SMDK6410: Set CFG1 1-3 On, CFG2 1-4 Off - */ - -/* - Playback (HeadPhone):- - $ amixer sset 'Headphone' unmute - $ amixer sset 'Right Headphone Out Mux' 'Headphone' - $ amixer sset 'Left Headphone Out Mux' 'Headphone' - $ amixer sset 'Right HP Mixer PCM' unmute - $ amixer sset 'Left HP Mixer PCM' unmute - - Capture (LineIn):- - $ amixer sset 'Right Capture Source' 'Line' - $ amixer sset 'Left Capture Source' 'Line' -*/ - -static struct snd_soc_dai_link smdk_dai = { - .name = "AC97", - .stream_name = "AC97 PCM", - .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], - .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], -}; - -static struct snd_soc_card smdk = { - .name = "SMDK", - .platform = &s3c24xx_soc_platform, - .dai_link = &smdk_dai, - .num_links = 1, -}; - -static struct snd_soc_device smdk_snd_ac97_devdata = { - .card = &smdk, - .codec_dev = &soc_codec_dev_wm9713, -}; - -static struct platform_device *smdk_snd_ac97_device; - -static int __init smdk_init(void) -{ - int ret; - - smdk_snd_ac97_device = platform_device_alloc("soc-audio", -1); - if (!smdk_snd_ac97_device) - return -ENOMEM; - - platform_set_drvdata(smdk_snd_ac97_device, - &smdk_snd_ac97_devdata); - smdk_snd_ac97_devdata.dev = &smdk_snd_ac97_device->dev; - - ret = platform_device_add(smdk_snd_ac97_device); - if (ret) - platform_device_put(smdk_snd_ac97_device); - - return ret; -} - -static void __exit smdk_exit(void) -{ - platform_device_unregister(smdk_snd_ac97_device); -} - -module_init(smdk_init); -module_exit(smdk_exit); - -/* Module information */ -MODULE_AUTHOR("Jaswinder Singh Brar, jassi.brar(a)samsung.com"); -MODULE_DESCRIPTION("ALSA SoC SMDK+WM9713"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig new file mode 100644 index 0000000..2a7cc22 --- /dev/null +++ b/sound/soc/samsung/Kconfig @@ -0,0 +1,127 @@ +config SND_S3C24XX_SOC + tristate "SoC Audio for the Samsung S3CXXXX chips" + depends on ARCH_S3C2410 || ARCH_S3C64XX + select S3C64XX_DMA if ARCH_S3C64XX + help + Say Y or M if you want to add support for codecs attached to + the S3C24XX AC97 or I2S interfaces. You will also need to + select the audio interfaces to support below. + +config SND_S3C24XX_SOC_I2S + tristate + select S3C2410_DMA + +config SND_S3C_I2SV2_SOC + tristate + +config SND_S3C2412_SOC_I2S + tristate + select SND_S3C_I2SV2_SOC + select S3C2410_DMA + +config SND_S3C64XX_SOC_I2S + tristate + select SND_S3C_I2SV2_SOC + select S3C64XX_DMA + +config SND_S3C64XX_SOC_I2S_V4 + tristate + select SND_S3C_I2SV2_SOC + select S3C64XX_DMA + +config SND_S3C_SOC_PCM + tristate + +config SND_S3C_SOC_AC97 + tristate + select SND_SOC_AC97_BUS + +config SND_S3C24XX_SOC_NEO1973_WM8753 + tristate "SoC I2S Audio support for NEO1973 - WM8753" + depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01 + select SND_S3C24XX_SOC_I2S + select SND_SOC_WM8753 + help + Say Y if you want to add support for SoC audio on smdk2440 + with the WM8753. + +config SND_S3C24XX_SOC_NEO1973_GTA02_WM8753 + tristate "Audio support for the Openmoko Neo FreeRunner (GTA02)" + depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA02 + select SND_S3C24XX_SOC_I2S + select SND_SOC_WM8753 + help + This driver provides audio support for the Openmoko Neo FreeRunner + smartphone. + +config SND_S3C24XX_SOC_JIVE_WM8750 + tristate "SoC I2S Audio support for Jive" + depends on SND_S3C24XX_SOC && MACH_JIVE + select SND_SOC_WM8750 + select SND_S3C2412_SOC_I2S + help + Sat Y if you want to add support for SoC audio on the Jive. + +config SND_S3C64XX_SOC_WM8580 + tristate "SoC I2S Audio support for WM8580 on SMDK64XX" + depends on SND_S3C24XX_SOC && MACH_SMDK6410 + select SND_SOC_WM8580 + select SND_S3C64XX_SOC_I2S_V4 + help + Say Y if you want to add support for SoC audio on the SMDK6410. + +config SND_S3C24XX_SOC_SMDK2443_WM9710 + tristate "SoC AC97 Audio support for SMDK2443 - WM9710" + depends on SND_S3C24XX_SOC && MACH_SMDK2443 + select S3C2410_DMA + select AC97_BUS + select SND_SOC_AC97_CODEC + select SND_S3C_SOC_AC97 + help + Say Y if you want to add support for SoC audio on smdk2443 + with the WM9710. + +config SND_S3C24XX_SOC_LN2440SBC_ALC650 + tristate "SoC AC97 Audio support for LN2440SBC - ALC650" + depends on SND_S3C24XX_SOC && ARCH_S3C2410 + select S3C2410_DMA + select AC97_BUS + select SND_SOC_AC97_CODEC + select SND_S3C_SOC_AC97 + help + Say Y if you want to add support for SoC audio on ln2440sbc + with the ALC650. + +config SND_S3C24XX_SOC_S3C24XX_UDA134X + tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" + depends on SND_S3C24XX_SOC && ARCH_S3C2410 + select SND_S3C24XX_SOC_I2S + select SND_SOC_L3 + select SND_SOC_UDA134X + +config SND_S3C24XX_SOC_SIMTEC + tristate + help + Internal node for common S3C24XX/Simtec suppor + +config SND_S3C24XX_SOC_SIMTEC_TLV320AIC23 + tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards" + depends on SND_S3C24XX_SOC && ARCH_S3C2410 + select SND_S3C24XX_SOC_I2S + select SND_SOC_TLV320AIC23 + select SND_S3C24XX_SOC_SIMTEC + +config SND_S3C24XX_SOC_SIMTEC_HERMES + tristate "SoC I2S Audio support for Simtec Hermes board" + depends on SND_S3C24XX_SOC && ARCH_S3C2410 + select SND_S3C24XX_SOC_I2S + select SND_SOC_TLV320AIC3X + select SND_S3C24XX_SOC_SIMTEC + +config SND_SOC_SMDK_WM9713 + tristate "SoC AC97 Audio support for SMDK with WM9713" + depends on SND_S3C24XX_SOC && MACH_SMDK6410 + select SND_SOC_WM9713 + select SND_S3C_SOC_AC97 + help + Sat Y if you want to add support for SoC audio on the SMDK. diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile new file mode 100644 index 0000000..81d8dc5 --- /dev/null +++ b/sound/soc/samsung/Makefile @@ -0,0 +1,43 @@ +# S3c24XX Platform Support +snd-soc-s3c24xx-objs := s3c-dma.o +snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o +snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o +snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o +snd-soc-s3c-ac97-objs := s3c-ac97.o +snd-soc-s3c64xx-i2s-v4-objs := s3c64xx-i2s-v4.o +snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o +snd-soc-s3c-pcm-objs := s3c-pcm.o + +obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o +obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o +obj-$(CONFIG_SND_S3C_SOC_AC97) += snd-soc-s3c-ac97.o +obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o +obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o +obj-$(CONFIG_SND_S3C64XX_SOC_I2S_V4) += snd-soc-s3c64xx-i2s-v4.o +obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o +obj-$(CONFIG_SND_S3C_SOC_PCM) += snd-soc-s3c-pcm.o + +# S3C24XX Machine Support +snd-soc-jive-wm8750-objs := jive_wm8750.o +snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o +snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o +snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o +snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o +snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o +snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o +snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o +snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o +snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o +snd-soc-smdk-wm9713-objs := smdk_wm9713.o + +obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o +obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o +obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o +obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o +obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o +obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o +obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o +obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c new file mode 100644 index 0000000..8c108b1 --- /dev/null +++ b/sound/soc/samsung/jive_wm8750.c @@ -0,0 +1,196 @@ +/* sound/soc/s3c24xx/jive_wm8750.c + * + * Copyright 2007,2008 Simtec Electronics + * + * Based on sound/soc/pxa/spitz.c + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand 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. +*/ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "s3c-dma.h" +#include "s3c2412-i2s.h" + +#include "../codecs/wm8750.h" + +static const struct snd_soc_dapm_route audio_map[] = { + { "Headphone Jack", NULL, "LOUT1" }, + { "Headphone Jack", NULL, "ROUT1" }, + { "Internal Speaker", NULL, "LOUT2" }, + { "Internal Speaker", NULL, "ROUT2" }, + { "LINPUT1", NULL, "Line Input" }, + { "RINPUT1", NULL, "Line Input" }, +}; + +static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Internal Speaker", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static int jive_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; + struct s3c_i2sv2_rate_calc div; + unsigned int clk = 0; + int ret = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + clk = 11289600; + break; + } + + s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), + s3c_i2sv2_get_clock(cpu_dai)); + + /* 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; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_RCLK, div.fs_div); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_PRESCALER, + div.clk_div - 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops jive_ops = { + .hw_params = jive_hw_params, +}; + +static int jive_wm8750_init(struct snd_soc_codec *codec) +{ + int err; + + /* These endpoints are not being used. */ + 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, "MONO"); + + /* Add jive specific widgets */ + err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, + ARRAY_SIZE(wm8750_dapm_widgets)); + if (err) { + printk(KERN_ERR "%s: failed to add widgets (%d)\n", + __func__, err); + return err; + } + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link jive_dai = { + .name = "wm8750", + .stream_name = "WM8750", + .cpu_dai = &s3c2412_i2s_dai, + .codec_dai = &wm8750_dai, + .init = jive_wm8750_init, + .ops = &jive_ops, +}; + +/* jive audio machine driver */ +static struct snd_soc_card snd_soc_machine_jive = { + .name = "Jive", + .platform = &s3c24xx_soc_platform, + .dai_link = &jive_dai, + .num_links = 1, +}; + +/* jive audio subsystem */ +static struct snd_soc_device jive_snd_devdata = { + .card = &snd_soc_machine_jive, + .codec_dev = &soc_codec_dev_wm8750, +}; + +static struct platform_device *jive_snd_device; + +static int __init jive_init(void) +{ + int ret; + + if (!machine_is_jive()) + return 0; + + printk("JIVE WM8750 Audio support\n"); + + jive_snd_device = platform_device_alloc("soc-audio", -1); + if (!jive_snd_device) + return -ENOMEM; + + platform_set_drvdata(jive_snd_device, &jive_snd_devdata); + jive_snd_devdata.dev = &jive_snd_device->dev; + ret = platform_device_add(jive_snd_device); + + if (ret) + platform_device_put(jive_snd_device); + + return ret; +} + +static void __exit jive_exit(void) +{ + platform_device_unregister(jive_snd_device); +} + +module_init(jive_init); +module_exit(jive_exit); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Jive Audio support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/lm4857.h b/sound/soc/samsung/lm4857.h new file mode 100644 index 0000000..0cf5b70 --- /dev/null +++ b/sound/soc/samsung/lm4857.h @@ -0,0 +1,32 @@ +/* + * lm4857.h -- ALSA Soc Audio Layer + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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 + * 18th Jun 2007 Initial version. + */ + +#ifndef LM4857_H_ +#define LM4857_H_ + +/* The register offsets in the cache array */ +#define LM4857_MVOL 0 +#define LM4857_LVOL 1 +#define LM4857_RVOL 2 +#define LM4857_CTRL 3 + +/* the shifts required to set these bits */ +#define LM4857_3D 5 +#define LM4857_WAKEUP 5 +#define LM4857_EPGAIN 4 + +#endif /*LM4857_H_*/ + diff --git a/sound/soc/samsung/ln2440sbc_alc650.c b/sound/soc/samsung/ln2440sbc_alc650.c new file mode 100644 index 0000000..ffa954f --- /dev/null +++ b/sound/soc/samsung/ln2440sbc_alc650.c @@ -0,0 +1,85 @@ +/* + * SoC audio for ln2440sbc + * + * Copyright 2007 KonekTel, a.s. + * Author: Ivan Kuten + * ivan.kuten(a)promwad.com + * + * Heavily based on smdk2443_wm9710.c + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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. + * + */ + +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/ac97.h" +#include "s3c-dma.h" +#include "s3c-ac97.h" + +static struct snd_soc_card ln2440sbc; + +static struct snd_soc_dai_link ln2440sbc_dai[] = { +{ + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], + .codec_dai = &ac97_dai, +}, +}; + +static struct snd_soc_card ln2440sbc = { + .name = "LN2440SBC", + .platform = &s3c24xx_soc_platform, + .dai_link = ln2440sbc_dai, + .num_links = ARRAY_SIZE(ln2440sbc_dai), +}; + +static struct snd_soc_device ln2440sbc_snd_ac97_devdata = { + .card = &ln2440sbc, + .codec_dev = &soc_codec_dev_ac97, +}; + +static struct platform_device *ln2440sbc_snd_ac97_device; + +static int __init ln2440sbc_init(void) +{ + int ret; + + ln2440sbc_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!ln2440sbc_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(ln2440sbc_snd_ac97_device, + &ln2440sbc_snd_ac97_devdata); + ln2440sbc_snd_ac97_devdata.dev = &ln2440sbc_snd_ac97_device->dev; + ret = platform_device_add(ln2440sbc_snd_ac97_device); + + if (ret) + platform_device_put(ln2440sbc_snd_ac97_device); + + return ret; +} + +static void __exit ln2440sbc_exit(void) +{ + platform_device_unregister(ln2440sbc_snd_ac97_device); +} + +module_init(ln2440sbc_init); +module_exit(ln2440sbc_exit); + +/* Module information */ +MODULE_AUTHOR("Ivan Kuten"); +MODULE_DESCRIPTION("ALSA SoC ALC650 LN2440SBC"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c new file mode 100644 index 0000000..209c259 --- /dev/null +++ b/sound/soc/samsung/neo1973_gta02_wm8753.c @@ -0,0 +1,506 @@ +/* + * neo1973_gta02_wm8753.c -- SoC audio for Openmoko Freerunner(GTA02) + * + * Copyright 2007 Openmoko Inc + * Author: Graeme Gregory <graeme(a)openmoko.org> + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory <linux(a)wolfsonmicro.com> + * Copyright 2009 Wolfson Microelectronics + * + * 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/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include <plat/regs-iis.h> + +#include <mach/regs-clock.h> +#include <asm/io.h> +#include <mach/gta02.h> +#include "../codecs/wm8753.h" +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" + +static struct snd_soc_card neo1973_gta02; + +static int neo1973_gta02_hifi_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; + unsigned int pll_out = 0, bclk = 0; + int ret = 0; + unsigned long iis_clkrate; + + iis_clkrate = s3c24xx_i2s_get_clockrate(); + + switch (params_rate(params)) { + case 8000: + case 16000: + pll_out = 12288000; + break; + case 48000: + bclk = WM8753_BCLK_DIV_4; + pll_out = 12288000; + break; + case 96000: + bclk = WM8753_BCLK_DIV_2; + pll_out = 12288000; + break; + case 11025: + bclk = WM8753_BCLK_DIV_16; + pll_out = 11289600; + break; + case 22050: + bclk = WM8753_BCLK_DIV_8; + pll_out = 11289600; + break; + case 44100: + bclk = WM8753_BCLK_DIV_4; + pll_out = 11289600; + break; + case 88200: + bclk = WM8753_BCLK_DIV_2; + pll_out = 11289600; + break; + } + + /* 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; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + /* set codec BCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(codec_dai, + WM8753_BCLKDIV, bclk); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(4, 4)); + if (ret < 0) + return ret; + + /* codec PLL input is PCLK/4 */ + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, + iis_clkrate / 4, pll_out); + if (ret < 0) + return ret; + + return 0; +} + +static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); +} + +/* + * Neo1973 WM8753 HiFi DAI opserations. + */ +static struct snd_soc_ops neo1973_gta02_hifi_ops = { + .hw_params = neo1973_gta02_hifi_hw_params, + .hw_free = neo1973_gta02_hifi_hw_free, +}; + +static int neo1973_gta02_voice_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; + unsigned int pcmdiv = 0; + int ret = 0; + unsigned long iis_clkrate; + + iis_clkrate = s3c24xx_i2s_get_clockrate(); + + if (params_rate(params) != 8000) + return -EINVAL; + if (params_channels(params) != 1) + return -EINVAL; + + pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ + + /* todo: gg check mode (DSP_B) against CSR datasheet */ + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, + 12288000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set codec PCM division for sample rate */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, + pcmdiv); + if (ret < 0) + return ret; + + /* configue and enable PLL for 12.288MHz output */ + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, + iis_clkrate / 4, 12288000); + if (ret < 0) + return ret; + + return 0; +} + +static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); +} + +static struct snd_soc_ops neo1973_gta02_voice_ops = { + .hw_params = neo1973_gta02_voice_hw_params, + .hw_free = neo1973_gta02_voice_hw_free, +}; + +#define LM4853_AMP 1 +#define LM4853_SPK 2 + +static u8 lm4853_state; + +/* This has no effect, it exists only to maintain compatibility with + * existing ALSA state files. + */ +static int lm4853_set_state(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int val = ucontrol->value.integer.value[0]; + + if (val) + lm4853_state |= LM4853_AMP; + else + lm4853_state &= ~LM4853_AMP; + + return 0; +} + +static int lm4853_get_state(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = lm4853_state & LM4853_AMP; + + return 0; +} + +static int lm4853_set_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int val = ucontrol->value.integer.value[0]; + + if (val) { + lm4853_state |= LM4853_SPK; + gpio_set_value(GTA02_GPIO_HP_IN, 0); + } else { + lm4853_state &= ~LM4853_SPK; + gpio_set_value(GTA02_GPIO_HP_IN, 1); + } + + return 0; +} + +static int lm4853_get_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = (lm4853_state & LM4853_SPK) >> 1; + + return 0; +} + +static int lm4853_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, + int event) +{ + gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(value)); + + return 0; +} + +static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Stereo Out", lm4853_event), + SND_SOC_DAPM_LINE("GSM Line Out", NULL), + SND_SOC_DAPM_LINE("GSM Line In", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Handset Mic", NULL), + SND_SOC_DAPM_SPK("Handset Spk", NULL), +}; + + +/* example machine audio_mapnections */ +static const struct snd_soc_dapm_route audio_map[] = { + + /* Connections to the lm4853 amp */ + {"Stereo Out", NULL, "LOUT1"}, + {"Stereo Out", NULL, "ROUT1"}, + + /* Connections to the GSM Module */ + {"GSM Line Out", NULL, "MONO1"}, + {"GSM Line Out", NULL, "MONO2"}, + {"RXP", NULL, "GSM Line In"}, + {"RXN", NULL, "GSM Line In"}, + + /* Connections to Headset */ + {"MIC1", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Headset Mic"}, + + /* Call Mic */ + {"MIC2", NULL, "Mic Bias"}, + {"MIC2N", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Handset Mic"}, + + /* Call Speaker */ + {"Handset Spk", NULL, "LOUT2"}, + {"Handset Spk", NULL, "ROUT2"}, + + /* Connect the ALC pins */ + {"ACIN", NULL, "ACOP"}, +}; + +static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = { + SOC_DAPM_PIN_SWITCH("Stereo Out"), + SOC_DAPM_PIN_SWITCH("GSM Line Out"), + SOC_DAPM_PIN_SWITCH("GSM Line In"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Handset Mic"), + SOC_DAPM_PIN_SWITCH("Handset Spk"), + + /* This has no effect, it exists only to maintain compatibility with + * existing ALSA state files. + */ + SOC_SINGLE_EXT("Amp State Switch", 6, 0, 1, 0, + lm4853_get_state, + lm4853_set_state), + SOC_SINGLE_EXT("Amp Spk Switch", 7, 0, 1, 0, + lm4853_get_spk, + lm4853_set_spk), +}; + +/* + * This is an example machine initialisation for a wm8753 connected to a + * neo1973 GTA02. + */ +static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec) +{ + int err; + + /* set up NC codec pins */ + 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 gta02 specific widgets */ + snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, + ARRAY_SIZE(wm8753_dapm_widgets)); + + /* add neo1973 gta02 specific controls */ + err = snd_soc_add_controls(codec, wm8753_neo1973_gta02_controls, + ARRAY_SIZE(wm8753_neo1973_gta02_controls)); + + if (err < 0) + return err; + + /* set up neo1973 gta02 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* set endpoints to default off mode */ + snd_soc_dapm_disable_pin(codec, "Stereo Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Handset Mic"); + snd_soc_dapm_disable_pin(codec, "Handset Spk"); + + /* allow audio paths from the GSM modem to run during suspend */ + snd_soc_dapm_ignore_suspend(codec, "Stereo Out"); + snd_soc_dapm_ignore_suspend(codec, "GSM Line Out"); + snd_soc_dapm_ignore_suspend(codec, "GSM Line In"); + snd_soc_dapm_ignore_suspend(codec, "Headset Mic"); + snd_soc_dapm_ignore_suspend(codec, "Handset Mic"); + snd_soc_dapm_ignore_suspend(codec, "Handset Spk"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* + * BT Codec DAI + */ +static struct snd_soc_dai bt_dai = { + .name = "Bluetooth", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_dai_link neo1973_gta02_dai[] = { +{ /* Hifi Playback - for similatious use with voice below */ + .name = "WM8753", + .stream_name = "WM8753 HiFi", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], + .init = neo1973_gta02_wm8753_init, + .ops = &neo1973_gta02_hifi_ops, +}, +{ /* Voice via BT */ + .name = "Bluetooth", + .stream_name = "Voice", + .cpu_dai = &bt_dai, + .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], + .ops = &neo1973_gta02_voice_ops, +}, +}; + +static struct snd_soc_card neo1973_gta02 = { + .name = "neo1973-gta02", + .platform = &s3c24xx_soc_platform, + .dai_link = neo1973_gta02_dai, + .num_links = ARRAY_SIZE(neo1973_gta02_dai), +}; + +static struct snd_soc_device neo1973_gta02_snd_devdata = { + .card = &neo1973_gta02, + .codec_dev = &soc_codec_dev_wm8753, +}; + +static struct platform_device *neo1973_gta02_snd_device; + +static int __init neo1973_gta02_init(void) +{ + int ret; + + if (!machine_is_neo1973_gta02()) { + printk(KERN_INFO + "Only GTA02 is supported by this ASoC driver\n"); + return -ENODEV; + } + + /* register bluetooth DAI here */ + ret = snd_soc_register_dai(&bt_dai); + if (ret) + return ret; + + neo1973_gta02_snd_device = platform_device_alloc("soc-audio", -1); + if (!neo1973_gta02_snd_device) + return -ENOMEM; + + platform_set_drvdata(neo1973_gta02_snd_device, + &neo1973_gta02_snd_devdata); + neo1973_gta02_snd_devdata.dev = &neo1973_gta02_snd_device->dev; + ret = platform_device_add(neo1973_gta02_snd_device); + + if (ret) { + platform_device_put(neo1973_gta02_snd_device); + return ret; + } + + /* Initialise GPIOs used by amp */ + ret = gpio_request(GTA02_GPIO_HP_IN, "GTA02_HP_IN"); + if (ret) { + pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_HP_IN); + goto err_unregister_device; + } + + ret = gpio_direction_output(GTA02_GPIO_AMP_HP_IN, 1); + if (ret) { + pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_HP_IN); + goto err_free_gpio_hp_in; + } + + ret = gpio_request(GTA02_GPIO_AMP_SHUT, "GTA02_AMP_SHUT"); + if (ret) { + pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_AMP_SHUT); + goto err_free_gpio_hp_in; + } + + ret = gpio_direction_output(GTA02_GPIO_AMP_SHUT, 1); + if (ret) { + pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_AMP_SHUT); + goto err_free_gpio_amp_shut; + } + + return 0; + +err_free_gpio_amp_shut: + gpio_free(GTA02_GPIO_AMP_SHUT); +err_free_gpio_hp_in: + gpio_free(GTA02_GPIO_HP_IN); +err_unregister_device: + platform_device_unregister(neo1973_gta02_snd_device); + return ret; +} +module_init(neo1973_gta02_init); + +static void __exit neo1973_gta02_exit(void) +{ + snd_soc_unregister_dai(&bt_dai); + platform_device_unregister(neo1973_gta02_snd_device); + gpio_free(GTA02_GPIO_HP_IN); + gpio_free(GTA02_GPIO_AMP_SHUT); +} +module_exit(neo1973_gta02_exit); + +/* Module information */ +MODULE_AUTHOR("Graeme Gregory, graeme(a)openmoko.org"); +MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 GTA02"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c new file mode 100644 index 0000000..0cb4f86 --- /dev/null +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -0,0 +1,707 @@ +/* + * neo1973_wm8753.c -- SoC audio for Neo1973 + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include <asm/mach-types.h> +#include <asm/hardware/scoop.h> +#include <mach/regs-clock.h> +#include <mach/regs-gpio.h> +#include <mach/hardware.h> +#include <linux/io.h> +#include <mach/spi-gpio.h> + +#include <plat/regs-iis.h> + +#include "../codecs/wm8753.h" +#include "lm4857.h" +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" + +/* define the scenarios */ +#define NEO_AUDIO_OFF 0 +#define NEO_GSM_CALL_AUDIO_HANDSET 1 +#define NEO_GSM_CALL_AUDIO_HEADSET 2 +#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3 +#define NEO_STEREO_TO_SPEAKERS 4 +#define NEO_STEREO_TO_HEADPHONES 5 +#define NEO_CAPTURE_HANDSET 6 +#define NEO_CAPTURE_HEADSET 7 +#define NEO_CAPTURE_BLUETOOTH 8 + +static struct snd_soc_card neo1973; +static struct i2c_client *i2c; + +static int neo1973_hifi_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; + unsigned int pll_out = 0, bclk = 0; + int ret = 0; + unsigned long iis_clkrate; + + pr_debug("Entered %s\n", __func__); + + iis_clkrate = s3c24xx_i2s_get_clockrate(); + + switch (params_rate(params)) { + case 8000: + case 16000: + pll_out = 12288000; + break; + case 48000: + bclk = WM8753_BCLK_DIV_4; + pll_out = 12288000; + break; + case 96000: + bclk = WM8753_BCLK_DIV_2; + pll_out = 12288000; + break; + case 11025: + bclk = WM8753_BCLK_DIV_16; + pll_out = 11289600; + break; + case 22050: + bclk = WM8753_BCLK_DIV_8; + pll_out = 11289600; + break; + case 44100: + bclk = WM8753_BCLK_DIV_4; + pll_out = 11289600; + break; + case 88200: + bclk = WM8753_BCLK_DIV_2; + pll_out = 11289600; + break; + } + + /* 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; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + /* set codec BCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(4, 4)); + if (ret < 0) + return ret; + + /* codec PLL input is PCLK/4 */ + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, + iis_clkrate / 4, pll_out); + if (ret < 0) + return ret; + + return 0; +} + +static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + pr_debug("Entered %s\n", __func__); + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); +} + +/* + * Neo1973 WM8753 HiFi DAI opserations. + */ +static struct snd_soc_ops neo1973_hifi_ops = { + .hw_params = neo1973_hifi_hw_params, + .hw_free = neo1973_hifi_hw_free, +}; + +static int neo1973_voice_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; + unsigned int pcmdiv = 0; + int ret = 0; + unsigned long iis_clkrate; + + pr_debug("Entered %s\n", __func__); + + iis_clkrate = s3c24xx_i2s_get_clockrate(); + + if (params_rate(params) != 8000) + return -EINVAL; + if (params_channels(params) != 1) + return -EINVAL; + + pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ + + /* todo: gg check mode (DSP_B) against CSR datasheet */ + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set codec PCM division for sample rate */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); + if (ret < 0) + return ret; + + /* configue and enable PLL for 12.288MHz output */ + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, + iis_clkrate / 4, 12288000); + if (ret < 0) + return ret; + + return 0; +} + +static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + pr_debug("Entered %s\n", __func__); + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); +} + +static struct snd_soc_ops neo1973_voice_ops = { + .hw_params = neo1973_voice_hw_params, + .hw_free = neo1973_voice_hw_free, +}; + +static int neo1973_scenario; + +static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = neo1973_scenario; + return 0; +} + +static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) +{ + pr_debug("Entered %s\n", __func__); + + switch (neo1973_scenario) { + case NEO_AUDIO_OFF: + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + case NEO_GSM_CALL_AUDIO_HANDSET: + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); + break; + case NEO_GSM_CALL_AUDIO_HEADSET: + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + case NEO_GSM_CALL_AUDIO_BLUETOOTH: + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + case NEO_STEREO_TO_SPEAKERS: + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + case NEO_STEREO_TO_HEADPHONES: + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + case NEO_CAPTURE_HANDSET: + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); + break; + case NEO_CAPTURE_HEADSET: + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + case NEO_CAPTURE_BLUETOOTH: + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + default: + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + } + + snd_soc_dapm_sync(codec); + + return 0; +} + +static int neo1973_set_scenario(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + pr_debug("Entered %s\n", __func__); + + if (neo1973_scenario == ucontrol->value.integer.value[0]) + return 0; + + neo1973_scenario = ucontrol->value.integer.value[0]; + set_scenario_endpoints(codec, neo1973_scenario); + return 1; +} + +static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; + +static void lm4857_write_regs(void) +{ + pr_debug("Entered %s\n", __func__); + + if (i2c_master_send(i2c, lm4857_regs, 4) != 4) + printk(KERN_ERR "lm4857: i2c write failed\n"); +} + +static int lm4857_get_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; + int shift = mc->shift; + int mask = mc->max; + + pr_debug("Entered %s\n", __func__); + + ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask; + return 0; +} + +static int lm4857_set_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; + int shift = mc->shift; + int mask = mc->max; + + if (((lm4857_regs[reg] >> shift) & mask) == + ucontrol->value.integer.value[0]) + return 0; + + lm4857_regs[reg] &= ~(mask << shift); + lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift; + lm4857_write_regs(); + return 1; +} + +static int lm4857_get_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = lm4857_regs[LM4857_CTRL] & 0x0F; + + pr_debug("Entered %s\n", __func__); + + if (value) + value -= 5; + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int lm4857_set_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = ucontrol->value.integer.value[0]; + + pr_debug("Entered %s\n", __func__); + + if (value) + value += 5; + + if ((lm4857_regs[LM4857_CTRL] & 0x0F) == value) + return 0; + + lm4857_regs[LM4857_CTRL] &= 0xF0; + lm4857_regs[LM4857_CTRL] |= value; + lm4857_write_regs(); + return 1; +} + +static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { + SND_SOC_DAPM_LINE("Audio Out", NULL), + SND_SOC_DAPM_LINE("GSM Line Out", NULL), + SND_SOC_DAPM_LINE("GSM Line In", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Call Mic", NULL), +}; + + +static const struct snd_soc_dapm_route dapm_routes[] = { + + /* Connections to the lm4857 amp */ + {"Audio Out", NULL, "LOUT1"}, + {"Audio Out", NULL, "ROUT1"}, + + /* Connections to the GSM Module */ + {"GSM Line Out", NULL, "MONO1"}, + {"GSM Line Out", NULL, "MONO2"}, + {"RXP", NULL, "GSM Line In"}, + {"RXN", NULL, "GSM Line In"}, + + /* Connections to Headset */ + {"MIC1", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Headset Mic"}, + + /* Call Mic */ + {"MIC2", NULL, "Mic Bias"}, + {"MIC2N", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Call Mic"}, + + /* Connect the ALC pins */ + {"ACIN", NULL, "ACOP"}, +}; + +static const char *lm4857_mode[] = { + "Off", + "Call Speaker", + "Stereo Speakers", + "Stereo Speakers + Headphones", + "Headphones" +}; + +static const struct soc_enum lm4857_mode_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode), +}; + +static const char *neo_scenarios[] = { + "Off", + "GSM Handset", + "GSM Headset", + "GSM Bluetooth", + "Speakers", + "Headphones", + "Capture Handset", + "Capture Headset", + "Capture Bluetooth" +}; + +static const struct soc_enum neo_scenario_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios), +}; + +static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); +static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); + +static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { + SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, stereo_tlv), + SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, stereo_tlv), + SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, mono_tlv), + SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], + lm4857_get_mode, lm4857_set_mode), + SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0], + neo1973_get_scenario, neo1973_set_scenario), + SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0, + lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0, + lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT("Amp Fast Wakeup Playback Switch", LM4857_CTRL, 5, 1, 0, + lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT("Amp Earpiece 6dB Playback Switch", LM4857_CTRL, 4, 1, 0, + lm4857_get_reg, lm4857_set_reg), +}; + +/* + * This is an example machine initialisation for a wm8753 connected to a + * neo1973 II. It is missing logic to detect hp/mic insertions and logic + * to re-route the audio in such an event. + */ +static int neo1973_wm8753_init(struct snd_soc_codec *codec) +{ + int err; + + pr_debug("Entered %s\n", __func__); + + /* set up NC codec pins */ + 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 */ + err = snd_soc_add_controls(codec, wm8753_neo1973_controls, + ARRAY_SIZE(8753_neo1973_controls)); + if (err < 0) + return err; + + /* set up neo1973 specific audio routes */ + err = snd_soc_dapm_add_routes(codec, dapm_routes, + ARRAY_SIZE(dapm_routes)); + + snd_soc_dapm_sync(codec); + return 0; +} + +/* + * BT Codec DAI + */ +static struct snd_soc_dai bt_dai = { + .name = "Bluetooth", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_dai_link neo1973_dai[] = { +{ /* Hifi Playback - for similatious use with voice below */ + .name = "WM8753", + .stream_name = "WM8753 HiFi", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], + .init = neo1973_wm8753_init, + .ops = &neo1973_hifi_ops, +}, +{ /* Voice via BT */ + .name = "Bluetooth", + .stream_name = "Voice", + .cpu_dai = &bt_dai, + .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], + .ops = &neo1973_voice_ops, +}, +}; + +static struct snd_soc_card neo1973 = { + .name = "neo1973", + .platform = &s3c24xx_soc_platform, + .dai_link = neo1973_dai, + .num_links = ARRAY_SIZE(neo1973_dai), +}; + +static struct snd_soc_device neo1973_snd_devdata = { + .card = &neo1973, + .codec_dev = &soc_codec_dev_wm8753, +}; + +static int lm4857_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + pr_debug("Entered %s\n", __func__); + + i2c = client; + + lm4857_write_regs(); + return 0; +} + +static int lm4857_i2c_remove(struct i2c_client *client) +{ + pr_debug("Entered %s\n", __func__); + + i2c = NULL; + + return 0; +} + +static u8 lm4857_state; + +static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) +{ + pr_debug("Entered %s\n", __func__); + + dev_dbg(&dev->dev, "lm4857_suspend\n"); + lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf; + if (lm4857_state) { + lm4857_regs[LM4857_CTRL] &= 0xf0; + lm4857_write_regs(); + } + return 0; +} + +static int lm4857_resume(struct i2c_client *dev) +{ + pr_debug("Entered %s\n", __func__); + + if (lm4857_state) { + lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f); + lm4857_write_regs(); + } + return 0; +} + +static void lm4857_shutdown(struct i2c_client *dev) +{ + pr_debug("Entered %s\n", __func__); + + dev_dbg(&dev->dev, "lm4857_shutdown\n"); + lm4857_regs[LM4857_CTRL] &= 0xf0; + lm4857_write_regs(); +} + +static const struct i2c_device_id lm4857_i2c_id[] = { + { "neo1973_lm4857", 0 }, + { } +}; + +static struct i2c_driver lm4857_i2c_driver = { + .driver = { + .name = "LM4857 I2C Amp", + .owner = THIS_MODULE, + }, + .suspend = lm4857_suspend, + .resume = lm4857_resume, + .shutdown = lm4857_shutdown, + .probe = lm4857_i2c_probe, + .remove = lm4857_i2c_remove, + .id_table = lm4857_i2c_id, +}; + +static struct platform_device *neo1973_snd_device; + +static int __init neo1973_init(void) +{ + int ret; + + pr_debug("Entered %s\n", __func__); + + if (!machine_is_neo1973_gta01()) { + printk(KERN_INFO + "Only GTA01 hardware supported by ASoC driver\n"); + return -ENODEV; + } + + neo1973_snd_device = platform_device_alloc("soc-audio", -1); + if (!neo1973_snd_device) + return -ENOMEM; + + platform_set_drvdata(neo1973_snd_device, &neo1973_snd_devdata); + neo1973_snd_devdata.dev = &neo1973_snd_device->dev; + ret = platform_device_add(neo1973_snd_device); + + if (ret) { + platform_device_put(neo1973_snd_device); + return ret; + } + + ret = i2c_add_driver(&lm4857_i2c_driver); + + if (ret != 0) + platform_device_unregister(neo1973_snd_device); + + return ret; +} + +static void __exit neo1973_exit(void) +{ + pr_debug("Entered %s\n", __func__); + + i2c_del_driver(&lm4857_i2c_driver); + platform_device_unregister(neo1973_snd_device); +} + +module_init(neo1973_init); +module_exit(neo1973_exit); + +/* Module information */ +MODULE_AUTHOR("Graeme Gregory, graeme(a)openmoko.org, www.openmoko.org"); +MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/regs-i2s-v2.h b/sound/soc/samsung/regs-i2s-v2.h new file mode 100644 index 0000000..5e5e568 --- /dev/null +++ b/sound/soc/samsung/regs-i2s-v2.h @@ -0,0 +1,115 @@ +/* linux/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h + * + * Copyright 2007 Simtec Electronics <linux(a)simtec.co.uk> + * http://armlinux.simtec.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 + * published by the Free Software Foundation. + * + * S3C2412 IIS register definition +*/ + +#ifndef __ASM_ARCH_REGS_S3C2412_IIS_H +#define __ASM_ARCH_REGS_S3C2412_IIS_H + +#define S3C2412_IISCON (0x00) +#define S3C2412_IISMOD (0x04) +#define S3C2412_IISFIC (0x08) +#define S3C2412_IISPSR (0x0C) +#define S3C2412_IISTXD (0x10) +#define S3C2412_IISRXD (0x14) + +#define S5PC1XX_IISFICS 0x18 +#define S5PC1XX_IISTXDS 0x1C + +#define S5PC1XX_IISCON_SW_RST (1 << 31) +#define S5PC1XX_IISCON_FRXOFSTATUS (1 << 26) +#define S5PC1XX_IISCON_FRXORINTEN (1 << 25) +#define S5PC1XX_IISCON_FTXSURSTAT (1 << 24) +#define S5PC1XX_IISCON_FTXSURINTEN (1 << 23) +#define S5PC1XX_IISCON_TXSDMAPAUSE (1 << 20) +#define S5PC1XX_IISCON_TXSDMACTIVE (1 << 18) + +#define S3C64XX_IISCON_FTXURSTATUS (1 << 17) +#define S3C64XX_IISCON_FTXURINTEN (1 << 16) +#define S3C64XX_IISCON_TXFIFO2_EMPTY (1 << 15) +#define S3C64XX_IISCON_TXFIFO1_EMPTY (1 << 14) +#define S3C64XX_IISCON_TXFIFO2_FULL (1 << 13) +#define S3C64XX_IISCON_TXFIFO1_FULL (1 << 12) + +#define S3C2412_IISCON_LRINDEX (1 << 11) +#define S3C2412_IISCON_TXFIFO_EMPTY (1 << 10) +#define S3C2412_IISCON_RXFIFO_EMPTY (1 << 9) +#define S3C2412_IISCON_TXFIFO_FULL (1 << 8) +#define S3C2412_IISCON_RXFIFO_FULL (1 << 7) +#define S3C2412_IISCON_TXDMA_PAUSE (1 << 6) +#define S3C2412_IISCON_RXDMA_PAUSE (1 << 5) +#define S3C2412_IISCON_TXCH_PAUSE (1 << 4) +#define S3C2412_IISCON_RXCH_PAUSE (1 << 3) +#define S3C2412_IISCON_TXDMA_ACTIVE (1 << 2) +#define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) +#define S3C2412_IISCON_IIS_ACTIVE (1 << 0) + +#define S5PC1XX_IISMOD_OPCLK_CDCLK_OUT (0 << 30) +#define S5PC1XX_IISMOD_OPCLK_CDCLK_IN (1 << 30) +#define S5PC1XX_IISMOD_OPCLK_BCLK_OUT (2 << 30) +#define S5PC1XX_IISMOD_OPCLK_PCLK (3 << 30) +#define S5PC1XX_IISMOD_OPCLK_MASK (3 << 30) +#define S5PC1XX_IISMOD_TXS_IDMA (1 << 28) /* Sec_TXFIFO use I-DMA */ +#define S5PC1XX_IISMOD_BLCS_MASK 0x3 +#define S5PC1XX_IISMOD_BLCS_SHIFT 26 +#define S5PC1XX_IISMOD_BLCP_MASK 0x3 +#define S5PC1XX_IISMOD_BLCP_SHIFT 24 + +#define S3C64XX_IISMOD_C2DD_HHALF (1 << 21) /* Discard Higher-half */ +#define S3C64XX_IISMOD_C2DD_LHALF (1 << 20) /* Discard Lower-half */ +#define S3C64XX_IISMOD_C1DD_HHALF (1 << 19) +#define S3C64XX_IISMOD_C1DD_LHALF (1 << 18) +#define S3C64XX_IISMOD_DC2_EN (1 << 17) +#define S3C64XX_IISMOD_DC1_EN (1 << 16) +#define S3C64XX_IISMOD_BLC_16BIT (0 << 13) +#define S3C64XX_IISMOD_BLC_8BIT (1 << 13) +#define S3C64XX_IISMOD_BLC_24BIT (2 << 13) +#define S3C64XX_IISMOD_BLC_MASK (3 << 13) + +#define S3C2412_IISMOD_IMS_SYSMUX (1 << 10) +#define S3C2412_IISMOD_SLAVE (1 << 11) +#define S3C2412_IISMOD_MODE_TXONLY (0 << 8) +#define S3C2412_IISMOD_MODE_RXONLY (1 << 8) +#define S3C2412_IISMOD_MODE_TXRX (2 << 8) +#define S3C2412_IISMOD_MODE_MASK (3 << 8) +#define S3C2412_IISMOD_LR_LLOW (0 << 7) +#define S3C2412_IISMOD_LR_RLOW (1 << 7) +#define S3C2412_IISMOD_SDF_IIS (0 << 5) +#define S3C2412_IISMOD_SDF_MSB (1 << 5) +#define S3C2412_IISMOD_SDF_LSB (2 << 5) +#define S3C2412_IISMOD_SDF_MASK (3 << 5) +#define S3C2412_IISMOD_RCLK_256FS (0 << 3) +#define S3C2412_IISMOD_RCLK_512FS (1 << 3) +#define S3C2412_IISMOD_RCLK_384FS (2 << 3) +#define S3C2412_IISMOD_RCLK_768FS (3 << 3) +#define S3C2412_IISMOD_RCLK_MASK (3 << 3) +#define S3C2412_IISMOD_BCLK_32FS (0 << 1) +#define S3C2412_IISMOD_BCLK_48FS (1 << 1) +#define S3C2412_IISMOD_BCLK_16FS (2 << 1) +#define S3C2412_IISMOD_BCLK_24FS (3 << 1) +#define S3C2412_IISMOD_BCLK_MASK (3 << 1) +#define S3C2412_IISMOD_8BIT (1 << 0) + +#define S3C64XX_IISMOD_CDCLKCON (1 << 12) + +#define S3C2412_IISPSR_PSREN (1 << 15) + +#define S3C64XX_IISFIC_TX2COUNT(x) (((x) >> 24) & 0xf) +#define S3C64XX_IISFIC_TX1COUNT(x) (((x) >> 16) & 0xf) + +#define S3C2412_IISFIC_TXFLUSH (1 << 15) +#define S3C2412_IISFIC_RXFLUSH (1 << 7) +#define S3C2412_IISFIC_TXCOUNT(x) (((x) >> 8) & 0xf) +#define S3C2412_IISFIC_RXCOUNT(x) (((x) >> 0) & 0xf) + +#define S5PC1XX_IISFICS_TXFLUSH (1 << 15) +#define S5PC1XX_IISFICS_TXCOUNT(x) (((x) >> 8) & 0x7f) + +#endif /* __ASM_ARCH_REGS_S3C2412_IIS_H */ diff --git a/sound/soc/samsung/s3c-ac97.c b/sound/soc/samsung/s3c-ac97.c new file mode 100644 index 0000000..ecf4fd0 --- /dev/null +++ b/sound/soc/samsung/s3c-ac97.c @@ -0,0 +1,521 @@ +/* sound/soc/s3c24xx/s3c-ac97.c + * + * ALSA SoC Audio Layer - S3C AC97 Controller driver + * Evolved from s3c2443-ac97.c + * + * Copyright (c) 2010 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * Credits: Graeme Gregory, Sean Choi + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/clk.h> + +#include <sound/soc.h> + +#include <plat/regs-ac97.h> +#include <mach/dma.h> +#include <plat/audio.h> + +#include "s3c-dma.h" +#include "s3c-ac97.h" + +#define AC_CMD_ADDR(x) (x << 16) +#define AC_CMD_DATA(x) (x & 0xffff) + +struct s3c_ac97_info { + unsigned state; + struct clk *ac97_clk; + void __iomem *regs; + struct mutex lock; + struct completion done; +}; +static struct s3c_ac97_info s3c_ac97; + +static struct s3c2410_dma_client s3c_dma_client_out = { + .name = "AC97 PCMOut" +}; + +static struct s3c2410_dma_client s3c_dma_client_in = { + .name = "AC97 PCMIn" +}; + +static struct s3c2410_dma_client s3c_dma_client_micin = { + .name = "AC97 MicIn" +}; + +static struct s3c_dma_params s3c_ac97_pcm_out = { + .client = &s3c_dma_client_out, + .dma_size = 4, +}; + +static struct s3c_dma_params s3c_ac97_pcm_in = { + .client = &s3c_dma_client_in, + .dma_size = 4, +}; + +static struct s3c_dma_params s3c_ac97_mic_in = { + .client = &s3c_dma_client_micin, + .dma_size = 4, +}; + +static void s3c_ac97_activate(struct snd_ac97 *ac97) +{ + u32 ac_glbctrl, stat; + + stat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT) & 0x7; + if (stat == S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE) + return; /* Return if already active */ + + INIT_COMPLETION(s3c_ac97.done); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) + printk(KERN_ERR "AC97: Unable to activate!"); +} + +static unsigned short s3c_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + u32 ac_glbctrl, ac_codec_cmd; + u32 stat, addr, data; + + mutex_lock(&s3c_ac97.lock); + + s3c_ac97_activate(ac97); + + INIT_COMPLETION(s3c_ac97.done); + + ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd = S3C_AC97_CODEC_CMD_READ | AC_CMD_ADDR(reg); + writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); + + udelay(50); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) + printk(KERN_ERR "AC97: Unable to read!"); + + stat = readl(s3c_ac97.regs + S3C_AC97_STAT); + addr = (stat >> 16) & 0x7f; + data = (stat & 0xffff); + + if (addr != reg) + printk(KERN_ERR "s3c-ac97: req addr = %02x, rep addr = %02x\n", reg, addr); + + mutex_unlock(&s3c_ac97.lock); + + return (unsigned short)data; +} + +static void s3c_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + u32 ac_glbctrl, ac_codec_cmd; + + mutex_lock(&s3c_ac97.lock); + + s3c_ac97_activate(ac97); + + INIT_COMPLETION(s3c_ac97.done); + + ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd = AC_CMD_ADDR(reg) | AC_CMD_DATA(val); + writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); + + udelay(50); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) + printk(KERN_ERR "AC97: Unable to write!"); + + ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; + writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); + + mutex_unlock(&s3c_ac97.lock); +} + +static void s3c_ac97_cold_reset(struct snd_ac97 *ac97) +{ + writel(S3C_AC97_GLBCTRL_COLDRESET, + s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + writel(0, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); +} + +static void s3c_ac97_warm_reset(struct snd_ac97 *ac97) +{ + u32 stat; + + stat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT) & 0x7; + if (stat == S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE) + return; /* Return if already active */ + + writel(S3C_AC97_GLBCTRL_WARMRESET, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + writel(0, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + s3c_ac97_activate(ac97); +} + +static irqreturn_t s3c_ac97_irq(int irq, void *dev_id) +{ + u32 ac_glbctrl, ac_glbstat; + + ac_glbstat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT); + + if (ac_glbstat & S3C_AC97_GLBSTAT_CODECREADY) { + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl &= ~S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + complete(&s3c_ac97.done); + } + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= (1<<30); /* Clear interrupt */ + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + return IRQ_HANDLED; +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = s3c_ac97_read, + .write = s3c_ac97_write, + .warm_reset = s3c_ac97_warm_reset, + .reset = s3c_ac97_cold_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct s3c_dma_params *dma_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = &s3c_ac97_pcm_out; + else + dma_data = &s3c_ac97_pcm_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + + return 0; +} + +static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + u32 ac_glbctrl; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK; + else + ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMOUTTM_MASK; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA; + else + ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + } + + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); + + return 0; +} + +static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return -ENODEV; + else + snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); + + return 0; +} + +static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + u32 ac_glbctrl; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ac_glbctrl |= S3C_AC97_GLBCTRL_MICINTM_DMA; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + } + + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); + + return 0; +} + +static struct snd_soc_dai_ops s3c_ac97_dai_ops = { + .hw_params = s3c_ac97_hw_params, + .trigger = s3c_ac97_trigger, +}; + +static struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { + .hw_params = s3c_ac97_hw_mic_params, + .trigger = s3c_ac97_mic_trigger, +}; + +struct snd_soc_dai s3c_ac97_dai[] = { + [S3C_AC97_DAI_PCM] = { + .name = "s3c-ac97", + .id = S3C_AC97_DAI_PCM, + .ac97_control = 1, + .playback = { + .stream_name = "AC97 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "AC97 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &s3c_ac97_dai_ops, + }, + [S3C_AC97_DAI_MIC] = { + .name = "s3c-ac97-mic", + .id = S3C_AC97_DAI_MIC, + .ac97_control = 1, + .capture = { + .stream_name = "AC97 Mic Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &s3c_ac97_mic_dai_ops, + }, +}; +EXPORT_SYMBOL_GPL(s3c_ac97_dai); + +static __devinit int s3c_ac97_probe(struct platform_device *pdev) +{ + struct resource *mem_res, *dmatx_res, *dmarx_res, *dmamic_res, *irq_res; + struct s3c_audio_pdata *ac97_pdata; + int ret; + + ac97_pdata = pdev->dev.platform_data; + if (!ac97_pdata || !ac97_pdata->cfg_gpio) { + dev_err(&pdev->dev, "cfg_gpio callback not provided!\n"); + return -EINVAL; + } + + /* Check for availability of necessary resource */ + dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmatx_res) { + dev_err(&pdev->dev, "Unable to get AC97-TX dma resource\n"); + return -ENXIO; + } + + dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmarx_res) { + dev_err(&pdev->dev, "Unable to get AC97-RX dma resource\n"); + return -ENXIO; + } + + dmamic_res = platform_get_resource(pdev, IORESOURCE_DMA, 2); + if (!dmamic_res) { + dev_err(&pdev->dev, "Unable to get AC97-MIC dma resource\n"); + return -ENXIO; + } + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem_res) { + dev_err(&pdev->dev, "Unable to get register resource\n"); + return -ENXIO; + } + + irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!irq_res) { + dev_err(&pdev->dev, "AC97 IRQ not provided!\n"); + return -ENXIO; + } + + if (!request_mem_region(mem_res->start, + resource_size(mem_res), "s3c-ac97")) { + dev_err(&pdev->dev, "Unable to request register region\n"); + return -EBUSY; + } + + s3c_ac97_pcm_out.channel = dmatx_res->start; + s3c_ac97_pcm_out.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; + s3c_ac97_pcm_in.channel = dmarx_res->start; + s3c_ac97_pcm_in.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; + s3c_ac97_mic_in.channel = dmamic_res->start; + s3c_ac97_mic_in.dma_addr = mem_res->start + S3C_AC97_MIC_DATA; + + init_completion(&s3c_ac97.done); + mutex_init(&s3c_ac97.lock); + + s3c_ac97.regs = ioremap(mem_res->start, resource_size(mem_res)); + if (s3c_ac97.regs == NULL) { + dev_err(&pdev->dev, "Unable to ioremap register region\n"); + ret = -ENXIO; + goto err1; + } + + s3c_ac97.ac97_clk = clk_get(&pdev->dev, "ac97"); + if (IS_ERR(s3c_ac97.ac97_clk)) { + dev_err(&pdev->dev, "s3c-ac97 failed to get ac97_clock\n"); + ret = -ENODEV; + goto err2; + } + clk_enable(s3c_ac97.ac97_clk); + + if (ac97_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + ret = -EINVAL; + goto err3; + } + + ret = request_irq(irq_res->start, s3c_ac97_irq, + IRQF_DISABLED, "AC97", NULL); + if (ret < 0) { + printk(KERN_ERR "s3c-ac97: interrupt request failed.\n"); + goto err4; + } + + s3c_ac97_dai[S3C_AC97_DAI_PCM].dev = &pdev->dev; + s3c_ac97_dai[S3C_AC97_DAI_MIC].dev = &pdev->dev; + + ret = snd_soc_register_dais(s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); + if (ret) + goto err5; + + return 0; + +err5: + free_irq(irq_res->start, NULL); +err4: +err3: + clk_disable(s3c_ac97.ac97_clk); + clk_put(s3c_ac97.ac97_clk); +err2: + iounmap(s3c_ac97.regs); +err1: + release_mem_region(mem_res->start, resource_size(mem_res)); + + return ret; +} + +static __devexit int s3c_ac97_remove(struct platform_device *pdev) +{ + struct resource *mem_res, *irq_res; + + snd_soc_unregister_dais(s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); + + irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (irq_res) + free_irq(irq_res->start, NULL); + + clk_disable(s3c_ac97.ac97_clk); + clk_put(s3c_ac97.ac97_clk); + + iounmap(s3c_ac97.regs); + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (mem_res) + release_mem_region(mem_res->start, resource_size(mem_res)); + + return 0; +} + +static struct platform_driver s3c_ac97_driver = { + .probe = s3c_ac97_probe, + .remove = s3c_ac97_remove, + .driver = { + .name = "s3c-ac97", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c_ac97_init(void) +{ + return platform_driver_register(&s3c_ac97_driver); +} +module_init(s3c_ac97_init); + +static void __exit s3c_ac97_exit(void) +{ + platform_driver_unregister(&s3c_ac97_driver); +} +module_exit(s3c_ac97_exit); + +MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_DESCRIPTION("AC97 driver for the Samsung SoC"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c-ac97.h b/sound/soc/samsung/s3c-ac97.h new file mode 100644 index 0000000..2781983 --- /dev/null +++ b/sound/soc/samsung/s3c-ac97.h @@ -0,0 +1,23 @@ +/* sound/soc/s3c24xx/s3c-ac97.h + * + * ALSA SoC Audio Layer - S3C AC97 Controller driver + * Evolved from s3c2443-ac97.h + * + * Copyright (c) 2010 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * Credits: Graeme Gregory, Sean Choi + * + * 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 __S3C_AC97_H_ +#define __S3C_AC97_H_ + +#define S3C_AC97_DAI_PCM 0 +#define S3C_AC97_DAI_MIC 1 + +extern struct snd_soc_dai s3c_ac97_dai[]; + +#endif /* __S3C_AC97_H_ */ diff --git a/sound/soc/samsung/s3c-dma.c b/sound/soc/samsung/s3c-dma.c new file mode 100644 index 0000000..1b61c23 --- /dev/null +++ b/sound/soc/samsung/s3c-dma.c @@ -0,0 +1,483 @@ +/* + * s3c-dma.c -- ALSA Soc Audio Layer + * + * (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * Copyright 2004-2005 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/dma.h> +#include <mach/hardware.h> +#include <mach/dma.h> + +#include "s3c-dma.h" + +static const struct snd_pcm_hardware s3c_dma_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 128*1024, + .period_bytes_min = PAGE_SIZE, + .period_bytes_max = PAGE_SIZE*2, + .periods_min = 2, + .periods_max = 128, + .fifo_size = 32, +}; + +struct s3c24xx_runtime_data { + spinlock_t lock; + int state; + unsigned int dma_loaded; + unsigned int dma_limit; + unsigned int dma_period; + dma_addr_t dma_start; + dma_addr_t dma_pos; + dma_addr_t dma_end; + struct s3c_dma_params *params; +}; + +/* s3c_dma_enqueue + * + * place a dma buffer onto the queue for the dma system + * to handle. +*/ +static void s3c_dma_enqueue(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + dma_addr_t pos = prtd->dma_pos; + unsigned int limit; + int ret; + + pr_debug("Entered %s\n", __func__); + + if (s3c_dma_has_circular()) + limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; + else + limit = prtd->dma_limit; + + pr_debug("%s: loaded %d, limit %d\n", + __func__, prtd->dma_loaded, limit); + + while (prtd->dma_loaded < limit) { + unsigned long len = prtd->dma_period; + + pr_debug("dma_loaded: %d\n", prtd->dma_loaded); + + if ((pos + len) > prtd->dma_end) { + len = prtd->dma_end - pos; + pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n", + __func__, len); + } + + ret = s3c2410_dma_enqueue(prtd->params->channel, + substream, pos, len); + + if (ret == 0) { + prtd->dma_loaded++; + pos += prtd->dma_period; + if (pos >= prtd->dma_end) + pos = prtd->dma_start; + } else + break; + } + + prtd->dma_pos = pos; +} + +static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, + void *dev_id, int size, + enum s3c2410_dma_buffresult result) +{ + struct snd_pcm_substream *substream = dev_id; + struct s3c24xx_runtime_data *prtd; + + pr_debug("Entered %s\n", __func__); + + if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR) + return; + + prtd = substream->runtime->private_data; + + if (substream) + snd_pcm_period_elapsed(substream); + + spin_lock(&prtd->lock); + if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { + prtd->dma_loaded--; + s3c_dma_enqueue(substream); + } + + spin_unlock(&prtd->lock); +} + +static int s3c_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned long totbytes = params_buffer_bytes(params); + struct s3c_dma_params *dma = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + int ret = 0; + + + pr_debug("Entered %s\n", __func__); + + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!dma) + return 0; + + /* this may get called several times by oss emulation + * with different params -HW */ + if (prtd->params == NULL) { + /* prepare DMA */ + prtd->params = dma; + + pr_debug("params %p, client %p, channel %d\n", prtd->params, + prtd->params->client, prtd->params->channel); + + ret = s3c2410_dma_request(prtd->params->channel, + prtd->params->client, NULL); + + if (ret < 0) { + printk(KERN_ERR "failed to get dma channel\n"); + return ret; + } + + /* use the circular buffering if we have it available. */ + if (s3c_dma_has_circular()) + s3c2410_dma_setflags(prtd->params->channel, + S3C2410_DMAF_CIRCULAR); + } + + s3c2410_dma_set_buffdone_fn(prtd->params->channel, + s3c24xx_audio_buffdone); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + runtime->dma_bytes = totbytes; + + spin_lock_irq(&prtd->lock); + prtd->dma_loaded = 0; + prtd->dma_limit = runtime->hw.periods_min; + prtd->dma_period = params_period_bytes(params); + prtd->dma_start = runtime->dma_addr; + prtd->dma_pos = prtd->dma_start; + prtd->dma_end = prtd->dma_start + totbytes; + spin_unlock_irq(&prtd->lock); + + return 0; +} + +static int s3c_dma_hw_free(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + + pr_debug("Entered %s\n", __func__); + + /* TODO - do we need to ensure DMA flushed */ + snd_pcm_set_runtime_buffer(substream, NULL); + + if (prtd->params) { + s3c2410_dma_free(prtd->params->channel, prtd->params->client); + prtd->params = NULL; + } + + return 0; +} + +static int s3c_dma_prepare(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!prtd->params) + return 0; + + /* channel needs configuring for mem=>device, increment memory addr, + * sync to pclk, half-word transfers to the IIS-FIFO. */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + s3c2410_dma_devconfig(prtd->params->channel, + S3C2410_DMASRC_MEM, + prtd->params->dma_addr); + } else { + s3c2410_dma_devconfig(prtd->params->channel, + S3C2410_DMASRC_HW, + prtd->params->dma_addr); + } + + s3c2410_dma_config(prtd->params->channel, + prtd->params->dma_size); + + /* flush the DMA channel */ + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH); + prtd->dma_loaded = 0; + prtd->dma_pos = prtd->dma_start; + + /* enqueue dma buffers */ + s3c_dma_enqueue(substream); + + return ret; +} + +static int s3c_dma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + prtd->state |= ST_RUNNING; + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + prtd->state &= ~ST_RUNNING; + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STOP); + break; + + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static snd_pcm_uframes_t +s3c_dma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd = runtime->private_data; + unsigned long res; + dma_addr_t src, dst; + + pr_debug("Entered %s\n", __func__); + + spin_lock(&prtd->lock); + s3c2410_dma_getposition(prtd->params->channel, &src, &dst); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + res = dst - prtd->dma_start; + else + res = src - prtd->dma_start; + + spin_unlock(&prtd->lock); + + pr_debug("Pointer %x %x\n", src, dst); + + /* we seem to be getting the odd error from the pcm library due + * to out-of-bounds pointers. this is maybe due to the dma engine + * not having loaded the new values for the channel before being + * callled... (todo - fix ) + */ + + if (res >= snd_pcm_lib_buffer_bytes(substream)) { + if (res == snd_pcm_lib_buffer_bytes(substream)) + res = 0; + } + + return bytes_to_frames(substream->runtime, res); +} + +static int s3c_dma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd; + + pr_debug("Entered %s\n", __func__); + + snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + snd_soc_set_runtime_hwparams(substream, &s3c_dma_hardware); + + prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + return 0; +} + +static int s3c_dma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd = runtime->private_data; + + pr_debug("Entered %s\n", __func__); + + if (!prtd) + pr_debug("s3c_dma_close called with prtd == NULL\n"); + + kfree(prtd); + + return 0; +} + +static int s3c_dma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + pr_debug("Entered %s\n", __func__); + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops s3c_dma_ops = { + .open = s3c_dma_open, + .close = s3c_dma_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = s3c_dma_hw_params, + .hw_free = s3c_dma_hw_free, + .prepare = s3c_dma_prepare, + .trigger = s3c_dma_trigger, + .pointer = s3c_dma_pointer, + .mmap = s3c_dma_mmap, +}; + +static int s3c_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = s3c_dma_hardware.buffer_bytes_max; + + pr_debug("Entered %s\n", __func__); + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + return 0; +} + +static void s3c_dma_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + pr_debug("Entered %s\n", __func__); + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static u64 s3c_dma_mask = DMA_BIT_MASK(32); + +static int s3c_dma_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + if (!card->dev->dma_mask) + card->dev->dma_mask = &s3c_dma_mask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = s3c_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + ret = s3c_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} + +struct snd_soc_platform s3c24xx_soc_platform = { + .name = "s3c24xx-audio", + .pcm_ops = &s3c_dma_ops, + .pcm_new = s3c_dma_new, + .pcm_free = s3c_dma_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); + +static int __init s3c24xx_soc_platform_init(void) +{ + return snd_soc_register_platform(&s3c24xx_soc_platform); +} +module_init(s3c24xx_soc_platform_init); + +static void __exit s3c24xx_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&s3c24xx_soc_platform); +} +module_exit(s3c24xx_soc_platform_exit); + +MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c-dma.h b/sound/soc/samsung/s3c-dma.h new file mode 100644 index 0000000..69bb6bf --- /dev/null +++ b/sound/soc/samsung/s3c-dma.h @@ -0,0 +1,31 @@ +/* + * s3c-dma.h -- + * + * 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. + * + * ALSA PCM interface for the Samsung S3C24xx CPU + */ + +#ifndef _S3C_AUDIO_H +#define _S3C_AUDIO_H + +#define ST_RUNNING (1<<0) +#define ST_OPENED (1<<1) + +struct s3c_dma_params { + struct s3c2410_dma_client *client; /* stream identifier */ + int channel; /* Channel ID */ + dma_addr_t dma_addr; + int dma_size; /* Size of the DMA transfer */ +}; + +#define S3C24XX_DAI_I2S 0 + +/* platform data */ +extern struct snd_soc_platform s3c24xx_soc_platform; +extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; + +#endif diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c new file mode 100644 index 0000000..13311c8 --- /dev/null +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -0,0 +1,776 @@ +/* sound/soc/s3c24xx/s3c-i2c-v2.c + * + * ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. + * + * Copyright (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory(a)wolfsonmicro.com + * linux(a)wolfsonmicro.com + * + * Copyright (c) 2008, 2007, 2004-2005 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/io.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/dma.h> + +#include "regs-i2s-v2.h" +#include "s3c-i2s-v2.h" +#include "s3c-dma.h" + +#undef S3C_IIS_V2_SUPPORTED + +#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) +#define S3C_IIS_V2_SUPPORTED +#endif + +#ifdef CONFIG_PLAT_S3C64XX +#define S3C_IIS_V2_SUPPORTED +#endif + +#ifndef S3C_IIS_V2_SUPPORTED +#error Unsupported CPU model +#endif + +#define S3C2412_I2S_DEBUG_CON 0 + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +#define bit_set(v, b) (((v) & (b)) ? 1 : 0) + +#if S3C2412_I2S_DEBUG_CON +static void dbg_showcon(const char *fn, u32 con) +{ + printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn, + bit_set(con, S3C2412_IISCON_LRINDEX), + bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY), + bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY), + bit_set(con, S3C2412_IISCON_TXFIFO_FULL), + bit_set(con, S3C2412_IISCON_RXFIFO_FULL)); + + printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n", + fn, + bit_set(con, S3C2412_IISCON_TXDMA_PAUSE), + bit_set(con, S3C2412_IISCON_RXDMA_PAUSE), + bit_set(con, S3C2412_IISCON_TXCH_PAUSE), + bit_set(con, S3C2412_IISCON_RXCH_PAUSE)); + printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn, + bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE), + bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE), + bit_set(con, S3C2412_IISCON_IIS_ACTIVE)); +} +#else +static inline void dbg_showcon(const char *fn, u32 con) +{ +} +#endif + + +/* Turn on or off the transmission path. */ +static void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) +{ + void __iomem *regs = i2s->regs; + u32 fic, con, mod; + + pr_debug("%s(%d)\n", __func__, on); + + fic = readl(regs + S3C2412_IISFIC); + con = readl(regs + S3C2412_IISCON); + mod = readl(regs + S3C2412_IISMOD); + + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); + + if (on) { + con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; + con &= ~S3C2412_IISCON_TXDMA_PAUSE; + con &= ~S3C2412_IISCON_TXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXONLY: + case S3C2412_IISMOD_MODE_TXRX: + /* do nothing, we are in the right mode */ + break; + + case S3C2412_IISMOD_MODE_RXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXRX; + break; + + default: + dev_err(i2s->dev, "TXEN: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); + break; + } + + writel(con, regs + S3C2412_IISCON); + writel(mod, regs + S3C2412_IISMOD); + } else { + /* Note, we do not have any indication that the FIFO problems + * tha the S3C2410/2440 had apply here, so we should be able + * to disable the DMA and TX without resetting the FIFOS. + */ + + con |= S3C2412_IISCON_TXDMA_PAUSE; + con |= S3C2412_IISCON_TXCH_PAUSE; + con &= ~S3C2412_IISCON_TXDMA_ACTIVE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXRX: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_RXONLY; + break; + + case S3C2412_IISMOD_MODE_TXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + con &= ~S3C2412_IISCON_IIS_ACTIVE; + break; + + default: + dev_err(i2s->dev, "TXDIS: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); + break; + } + + writel(mod, regs + S3C2412_IISMOD); + writel(con, regs + S3C2412_IISCON); + } + + fic = readl(regs + S3C2412_IISFIC); + dbg_showcon(__func__, con); + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); +} + +static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) +{ + void __iomem *regs = i2s->regs; + u32 fic, con, mod; + + pr_debug("%s(%d)\n", __func__, on); + + fic = readl(regs + S3C2412_IISFIC); + con = readl(regs + S3C2412_IISCON); + mod = readl(regs + S3C2412_IISMOD); + + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); + + if (on) { + con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; + con &= ~S3C2412_IISCON_RXDMA_PAUSE; + con &= ~S3C2412_IISCON_RXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXRX: + case S3C2412_IISMOD_MODE_RXONLY: + /* do nothing, we are in the right mode */ + break; + + case S3C2412_IISMOD_MODE_TXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXRX; + break; + + default: + dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); + } + + writel(mod, regs + S3C2412_IISMOD); + writel(con, regs + S3C2412_IISCON); + } else { + /* See txctrl notes on FIFOs. */ + + con &= ~S3C2412_IISCON_RXDMA_ACTIVE; + con |= S3C2412_IISCON_RXDMA_PAUSE; + con |= S3C2412_IISCON_RXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_RXONLY: + con &= ~S3C2412_IISCON_IIS_ACTIVE; + mod &= ~S3C2412_IISMOD_MODE_MASK; + break; + + case S3C2412_IISMOD_MODE_TXRX: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXONLY; + break; + + default: + dev_err(i2s->dev, "RXDIS: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); + } + + writel(con, regs + S3C2412_IISCON); + writel(mod, regs + S3C2412_IISMOD); + } + + fic = readl(regs + S3C2412_IISFIC); + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); +} + +#define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t) + +/* + * Wait for the LR signal to allow synchronisation to the L/R clock + * from the codec. May only be needed for slave mode. + */ +static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s) +{ + u32 iiscon; + unsigned long loops = msecs_to_loops(5); + + pr_debug("Entered %s\n", __func__); + + while (--loops) { + iiscon = readl(i2s->regs + S3C2412_IISCON); + if (iiscon & S3C2412_IISCON_LRINDEX) + break; + + cpu_relax(); + } + + if (!loops) { + printk(KERN_ERR "%s: timeout\n", __func__); + return -ETIMEDOUT; + } + + return 0; +} + +/* + * Set S3C2412 I2S DAI format + */ +static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + iismod = readl(i2s->regs + S3C2412_IISMOD); + pr_debug("hw_params r: IISMOD: %x \n", iismod); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + i2s->master = 0; + iismod |= S3C2412_IISMOD_SLAVE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + i2s->master = 1; + iismod &= ~S3C2412_IISMOD_SLAVE; + break; + default: + pr_err("unknwon master/slave format\n"); + return -EINVAL; + } + + iismod &= ~S3C2412_IISMOD_SDF_MASK; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + iismod |= S3C2412_IISMOD_LR_RLOW; + iismod |= S3C2412_IISMOD_SDF_MSB; + break; + case SND_SOC_DAIFMT_LEFT_J: + iismod |= S3C2412_IISMOD_LR_RLOW; + iismod |= S3C2412_IISMOD_SDF_LSB; + break; + case SND_SOC_DAIFMT_I2S: + iismod &= ~S3C2412_IISMOD_LR_RLOW; + iismod |= S3C2412_IISMOD_SDF_IIS; + break; + default: + pr_err("Unknown data format\n"); + return -EINVAL; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + pr_debug("hw_params w: IISMOD: %x \n", iismod); + return 0; +} + +static int s3c_i2sv2_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *socdai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *dai = rtd->dai; + struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai); + struct s3c_dma_params *dma_data; + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = i2s->dma_playback; + else + dma_data = i2s->dma_capture; + + snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); + + /* Working copies of register */ + iismod = readl(i2s->regs + S3C2412_IISMOD); + pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); + + iismod &= ~S3C64XX_IISMOD_BLC_MASK; + /* Sample size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod |= S3C64XX_IISMOD_BLC_8BIT; + break; + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S24_LE: + iismod |= S3C64XX_IISMOD_BLC_24BIT; + break; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); + + return 0; +} + +static int s3c_i2sv2_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 iismod = readl(i2s->regs + S3C2412_IISMOD); + + pr_debug("Entered %s\n", __func__); + pr_debug("%s r: IISMOD: %x\n", __func__, iismod); + + switch (clk_id) { + case S3C_I2SV2_CLKSRC_PCLK: + iismod &= ~S3C2412_IISMOD_IMS_SYSMUX; + break; + + case S3C_I2SV2_CLKSRC_AUDIOBUS: + iismod |= S3C2412_IISMOD_IMS_SYSMUX; + break; + + case S3C_I2SV2_CLKSRC_CDCLK: + /* Error if controller doesn't have the CDCLKCON bit */ + if (!(i2s->feature & S3C_FEATURE_CDCLKCON)) + return -EINVAL; + + switch (dir) { + case SND_SOC_CLOCK_IN: + iismod |= S3C64XX_IISMOD_CDCLKCON; + break; + case SND_SOC_CLOCK_OUT: + iismod &= ~S3C64XX_IISMOD_CDCLKCON; + break; + default: + return -EINVAL; + } + break; + + default: + return -EINVAL; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + pr_debug("%s w: IISMOD: %x\n", __func__, iismod); + + return 0; +} + +static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_i2sv2_info *i2s = to_info(rtd->dai->cpu_dai); + int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + unsigned long irqs; + int ret = 0; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + + pr_debug("Entered %s\n", __func__); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* On start, ensure that the FIFOs are cleared and reset. */ + + writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH, + i2s->regs + S3C2412_IISFIC); + + /* clear again, just in case */ + writel(0x0, i2s->regs + S3C2412_IISFIC); + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!i2s->master) { + ret = s3c2412_snd_lrsync(i2s); + if (ret) + goto exit_err; + } + + local_irq_save(irqs); + + if (capture) + s3c2412_snd_rxctrl(i2s, 1); + else + s3c2412_snd_txctrl(i2s, 1); + + local_irq_restore(irqs); + + /* + * Load the next buffer to DMA to meet the reqirement + * of the auto reload mechanism of S3C24XX. + * This call won't bother S3C64XX. + */ + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + local_irq_save(irqs); + + if (capture) + s3c2412_snd_rxctrl(i2s, 0); + else + s3c2412_snd_txctrl(i2s, 0); + + local_irq_restore(irqs); + break; + default: + ret = -EINVAL; + break; + } + +exit_err: + return ret; +} + +/* + * Set S3C2412 Clock dividers + */ +static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 reg; + + pr_debug("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div); + + switch (div_id) { + case S3C_I2SV2_DIV_BCLK: + switch (div) { + case 16: + div = S3C2412_IISMOD_BCLK_16FS; + break; + + case 32: + div = S3C2412_IISMOD_BCLK_32FS; + break; + + case 24: + div = S3C2412_IISMOD_BCLK_24FS; + break; + + case 48: + div = S3C2412_IISMOD_BCLK_48FS; + break; + + default: + return -EINVAL; + } + + reg = readl(i2s->regs + S3C2412_IISMOD); + reg &= ~S3C2412_IISMOD_BCLK_MASK; + writel(reg | div, i2s->regs + S3C2412_IISMOD); + + pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); + break; + + case S3C_I2SV2_DIV_RCLK: + switch (div) { + case 256: + div = S3C2412_IISMOD_RCLK_256FS; + break; + + case 384: + div = S3C2412_IISMOD_RCLK_384FS; + break; + + case 512: + div = S3C2412_IISMOD_RCLK_512FS; + break; + + case 768: + div = S3C2412_IISMOD_RCLK_768FS; + break; + + default: + return -EINVAL; + } + + reg = readl(i2s->regs + S3C2412_IISMOD); + reg &= ~S3C2412_IISMOD_RCLK_MASK; + writel(reg | div, i2s->regs + S3C2412_IISMOD); + pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); + break; + + case S3C_I2SV2_DIV_PRESCALER: + if (div >= 0) { + writel((div << 8) | S3C2412_IISPSR_PSREN, + i2s->regs + S3C2412_IISPSR); + } else { + writel(0x0, i2s->regs + S3C2412_IISPSR); + } + pr_debug("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR)); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_sframes_t s3c2412_i2s_delay(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + u32 reg = readl(i2s->regs + S3C2412_IISFIC); + snd_pcm_sframes_t delay; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + delay = S3C2412_IISFIC_TXCOUNT(reg); + else + delay = S3C2412_IISFIC_RXCOUNT(reg); + + return delay; +} + +struct clk *s3c_i2sv2_get_clock(struct snd_soc_dai *cpu_dai) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 iismod = readl(i2s->regs + S3C2412_IISMOD); + + if (iismod & S3C2412_IISMOD_IMS_SYSMUX) + return i2s->iis_cclk; + else + return i2s->iis_pclk; +} +EXPORT_SYMBOL_GPL(s3c_i2sv2_get_clock); + +/* default table of all avaialable root fs divisors */ +static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 }; + +int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, + unsigned int *fstab, + unsigned int rate, struct clk *clk) +{ + unsigned long clkrate = clk_get_rate(clk); + unsigned int div; + unsigned int fsclk; + unsigned int actual; + unsigned int fs; + unsigned int fsdiv; + signed int deviation = 0; + unsigned int best_fs = 0; + unsigned int best_div = 0; + unsigned int best_rate = 0; + unsigned int best_deviation = INT_MAX; + + pr_debug("Input clock rate %ldHz\n", clkrate); + + if (fstab == NULL) + fstab = iis_fs_tab; + + for (fs = 0; fs < ARRAY_SIZE(iis_fs_tab); fs++) { + fsdiv = iis_fs_tab[fs]; + + fsclk = clkrate / fsdiv; + div = fsclk / rate; + + if ((fsclk % rate) > (rate / 2)) + div++; + + if (div <= 1) + continue; + + actual = clkrate / (fsdiv * div); + deviation = actual - rate; + + printk(KERN_DEBUG "%ufs: div %u => result %u, deviation %d\n", + fsdiv, div, actual, deviation); + + deviation = abs(deviation); + + if (deviation < best_deviation) { + best_fs = fsdiv; + best_div = div; + best_rate = actual; + best_deviation = deviation; + } + + if (deviation == 0) + break; + } + + printk(KERN_DEBUG "best: fs=%u, div=%u, rate=%u\n", + best_fs, best_div, best_rate); + + info->fs_div = best_fs; + info->clk_div = best_div; + + return 0; +} +EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate); + +int s3c_i2sv2_probe(struct platform_device *pdev, + struct snd_soc_dai *dai, + struct s3c_i2sv2_info *i2s, + unsigned long base) +{ + struct device *dev = &pdev->dev; + unsigned int iismod; + + i2s->dev = dev; + + /* record our i2s structure for later use in the callbacks */ + dai->private_data = i2s; + + if (!base) { + struct resource *res = platform_get_resource(pdev, + IORESOURCE_MEM, + 0); + if (!res) { + dev_err(dev, "Unable to get register resource\n"); + return -ENXIO; + } + + if (!request_mem_region(res->start, resource_size(res), + "s3c64xx-i2s-v4")) { + dev_err(dev, "Unable to request register region\n"); + return -EBUSY; + } + + base = res->start; + } + + i2s->regs = ioremap(base, 0x100); + if (i2s->regs == NULL) { + dev_err(dev, "cannot ioremap registers\n"); + return -ENXIO; + } + + i2s->iis_pclk = clk_get(dev, "iis"); + if (IS_ERR(i2s->iis_pclk)) { + dev_err(dev, "failed to get iis_clock\n"); + iounmap(i2s->regs); + return -ENOENT; + } + + clk_enable(i2s->iis_pclk); + + /* Mark ourselves as in TXRX mode so we can run through our cleanup + * process without warnings. */ + iismod = readl(i2s->regs + S3C2412_IISMOD); + iismod |= S3C2412_IISMOD_MODE_TXRX; + writel(iismod, i2s->regs + S3C2412_IISMOD); + s3c2412_snd_txctrl(i2s, 0); + s3c2412_snd_rxctrl(i2s, 0); + + return 0; +} +EXPORT_SYMBOL_GPL(s3c_i2sv2_probe); + +#ifdef CONFIG_PM +static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + u32 iismod; + + if (dai->active) { + i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); + i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); + i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); + + /* some basic suspend checks */ + + iismod = readl(i2s->regs + S3C2412_IISMOD); + + if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) + pr_warning("%s: RXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) + pr_warning("%s: TXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_IIS_ACTIVE) + pr_warning("%s: IIS active\n", __func__); + } + + return 0; +} + +static int s3c2412_i2s_resume(struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + + pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n", + dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); + + if (dai->active) { + writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); + writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); + writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); + + writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, + i2s->regs + S3C2412_IISFIC); + + ndelay(250); + writel(0x0, i2s->regs + S3C2412_IISFIC); + } + + return 0; +} +#else +#define s3c2412_i2s_suspend NULL +#define s3c2412_i2s_resume NULL +#endif + +int s3c_i2sv2_register_dai(struct snd_soc_dai *dai) +{ + struct snd_soc_dai_ops *ops = dai->ops; + + ops->trigger = s3c2412_i2s_trigger; + if (!ops->hw_params) + ops->hw_params = s3c_i2sv2_hw_params; + ops->set_fmt = s3c2412_i2s_set_fmt; + ops->set_clkdiv = s3c2412_i2s_set_clkdiv; + ops->set_sysclk = s3c_i2sv2_set_sysclk; + + /* Allow overriding by (for example) IISv4 */ + if (!ops->delay) + ops->delay = s3c2412_i2s_delay; + + dai->suspend = s3c2412_i2s_suspend; + dai->resume = s3c2412_i2s_resume; + + return snd_soc_register_dai(dai); +} +EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai); + +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c-i2s-v2.h b/sound/soc/samsung/s3c-i2s-v2.h new file mode 100644 index 0000000..766f43a --- /dev/null +++ b/sound/soc/samsung/s3c-i2s-v2.h @@ -0,0 +1,103 @@ +/* sound/soc/s3c24xx/s3c-i2s-v2.h + * + * ALSA Soc Audio Layer - S3C_I2SV2 I2S driver + * + * Copyright (c) 2007 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. +*/ + +/* This code is the core support for the I2S block found in a number of + * Samsung SoC devices which is unofficially named I2S-V2. Currently the + * S3C2412 and the S3C64XX series use this block to provide 1 or 2 I2S + * channels via configurable GPIO. + */ + +#ifndef __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H +#define __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H __FILE__ + +#define S3C_I2SV2_DIV_BCLK (1) +#define S3C_I2SV2_DIV_RCLK (2) +#define S3C_I2SV2_DIV_PRESCALER (3) + +#define S3C_I2SV2_CLKSRC_PCLK 0 +#define S3C_I2SV2_CLKSRC_AUDIOBUS 1 +#define S3C_I2SV2_CLKSRC_CDCLK 2 + +/* Set this flag for I2S controllers that have the bit IISMOD[12] + * bridge/break RCLK signal and external Xi2sCDCLK pin. + */ +#define S3C_FEATURE_CDCLKCON (1 << 0) + +/** + * struct s3c_i2sv2_info - S3C I2S-V2 information + * @dev: The parent device passed to use from the probe. + * @regs: The pointer to the device registe block. + * @feature: Set of bit-flags indicating features of the controller. + * @master: True if the I2S core is the I2S bit clock master. + * @dma_playback: DMA information for playback channel. + * @dma_capture: DMA information for capture channel. + * @suspend_iismod: PM save for the IISMOD register. + * @suspend_iiscon: PM save for the IISCON register. + * @suspend_iispsr: PM save for the IISPSR register. + * + * This is the private codec state for the hardware associated with an + * I2S channel such as the register mappings and clock sources. + */ +struct s3c_i2sv2_info { + struct device *dev; + void __iomem *regs; + + u32 feature; + + struct clk *iis_pclk; + struct clk *iis_cclk; + + unsigned char master; + + struct s3c_dma_params *dma_playback; + struct s3c_dma_params *dma_capture; + + u32 suspend_iismod; + u32 suspend_iiscon; + u32 suspend_iispsr; +}; + +extern struct clk *s3c_i2sv2_get_clock(struct snd_soc_dai *cpu_dai); + +struct s3c_i2sv2_rate_calc { + unsigned int clk_div; /* for prescaler */ + unsigned int fs_div; /* for root frame clock */ +}; + +extern int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, + unsigned int *fstab, + unsigned int rate, struct clk *clk); + +/** + * s3c_i2sv2_probe - probe for i2s device helper + * @pdev: The platform device supplied to the original probe. + * @dai: The ASoC DAI structure supplied to the original probe. + * @i2s: Our local i2s structure to fill in. + * @base: The base address for the registers. + */ +extern int s3c_i2sv2_probe(struct platform_device *pdev, + struct snd_soc_dai *dai, + struct s3c_i2sv2_info *i2s, + unsigned long base); + +/** + * s3c_i2sv2_register_dai - register dai with soc core + * @dai: The snd_soc_dai structure to register + * + * Fill in any missing fields and then register the given dai with the + * soc core. + */ +extern int s3c_i2sv2_register_dai(struct snd_soc_dai *dai); + +#endif /* __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H */ diff --git a/sound/soc/samsung/s3c-pcm.c b/sound/soc/samsung/s3c-pcm.c new file mode 100644 index 0000000..326f0a9 --- /dev/null +++ b/sound/soc/samsung/s3c-pcm.c @@ -0,0 +1,554 @@ +/* sound/soc/s3c24xx/s3c-pcm.c + * + * ALSA SoC Audio Layer - S3C PCM-Controller driver + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * based upon I2S drivers by Ben Dooks. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/kernel.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <plat/audio.h> +#include <plat/dma.h> + +#include "s3c-dma.h" +#include "s3c-pcm.h" + +static struct s3c2410_dma_client s3c_pcm_dma_client_out = { + .name = "PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c_pcm_dma_client_in = { + .name = "PCM Stereo in" +}; + +static struct s3c_dma_params s3c_pcm_stereo_out[] = { + [0] = { + .client = &s3c_pcm_dma_client_out, + .dma_size = 4, + }, + [1] = { + .client = &s3c_pcm_dma_client_out, + .dma_size = 4, + }, +}; + +static struct s3c_dma_params s3c_pcm_stereo_in[] = { + [0] = { + .client = &s3c_pcm_dma_client_in, + .dma_size = 4, + }, + [1] = { + .client = &s3c_pcm_dma_client_in, + .dma_size = 4, + }, +}; + +static struct s3c_pcm_info s3c_pcm[2]; + +static inline struct s3c_pcm_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static void s3c_pcm_snd_txctrl(struct s3c_pcm_info *pcm, int on) +{ + void __iomem *regs = pcm->regs; + u32 ctl, clkctl; + + clkctl = readl(regs + S3C_PCM_CLKCTL); + ctl = readl(regs + S3C_PCM_CTL); + ctl &= ~(S3C_PCM_CTL_TXDIPSTICK_MASK + << S3C_PCM_CTL_TXDIPSTICK_SHIFT); + + if (on) { + ctl |= S3C_PCM_CTL_TXDMA_EN; + ctl |= S3C_PCM_CTL_TXFIFO_EN; + ctl |= S3C_PCM_CTL_ENABLE; + ctl |= (0x20<<S3C_PCM_CTL_TXDIPSTICK_SHIFT); + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } else { + ctl &= ~S3C_PCM_CTL_TXDMA_EN; + ctl &= ~S3C_PCM_CTL_TXFIFO_EN; + + if (!(ctl & S3C_PCM_CTL_RXFIFO_EN)) { + ctl &= ~S3C_PCM_CTL_ENABLE; + if (!pcm->idleclk) + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } + } + + writel(clkctl, regs + S3C_PCM_CLKCTL); + writel(ctl, regs + S3C_PCM_CTL); +} + +static void s3c_pcm_snd_rxctrl(struct s3c_pcm_info *pcm, int on) +{ + void __iomem *regs = pcm->regs; + u32 ctl, clkctl; + + ctl = readl(regs + S3C_PCM_CTL); + clkctl = readl(regs + S3C_PCM_CLKCTL); + + if (on) { + ctl |= S3C_PCM_CTL_RXDMA_EN; + ctl |= S3C_PCM_CTL_RXFIFO_EN; + ctl |= S3C_PCM_CTL_ENABLE; + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } else { + ctl &= ~S3C_PCM_CTL_RXDMA_EN; + ctl &= ~S3C_PCM_CTL_RXFIFO_EN; + + if (!(ctl & S3C_PCM_CTL_TXFIFO_EN)) { + ctl &= ~S3C_PCM_CTL_ENABLE; + if (!pcm->idleclk) + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } + } + + writel(clkctl, regs + S3C_PCM_CLKCTL); + writel(ctl, regs + S3C_PCM_CTL); +} + +static int s3c_pcm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_pcm_info *pcm = to_info(rtd->dai->cpu_dai); + unsigned long flags; + + dev_dbg(pcm->dev, "Entered %s\n", __func__); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&pcm->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + s3c_pcm_snd_rxctrl(pcm, 1); + else + s3c_pcm_snd_txctrl(pcm, 1); + + spin_unlock_irqrestore(&pcm->lock, flags); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&pcm->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + s3c_pcm_snd_rxctrl(pcm, 0); + else + s3c_pcm_snd_txctrl(pcm, 0); + + spin_unlock_irqrestore(&pcm->lock, flags); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *socdai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *dai = rtd->dai; + struct s3c_pcm_info *pcm = to_info(dai->cpu_dai); + struct s3c_dma_params *dma_data; + void __iomem *regs = pcm->regs; + struct clk *clk; + int sclk_div, sync_div; + unsigned long flags; + u32 clkctl; + + dev_dbg(pcm->dev, "Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = pcm->dma_playback; + else + dma_data = pcm->dma_capture; + + snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); + + /* Strictly check for sample size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + default: + return -EINVAL; + } + + spin_lock_irqsave(&pcm->lock, flags); + + /* Get hold of the PCMSOURCE_CLK */ + clkctl = readl(regs + S3C_PCM_CLKCTL); + if (clkctl & S3C_PCM_CLKCTL_SERCLKSEL_PCLK) + clk = pcm->pclk; + else + clk = pcm->cclk; + + /* Set the SCLK divider */ + sclk_div = clk_get_rate(clk) / pcm->sclk_per_fs / + params_rate(params) / 2 - 1; + + clkctl &= ~(S3C_PCM_CLKCTL_SCLKDIV_MASK + << S3C_PCM_CLKCTL_SCLKDIV_SHIFT); + clkctl |= ((sclk_div & S3C_PCM_CLKCTL_SCLKDIV_MASK) + << S3C_PCM_CLKCTL_SCLKDIV_SHIFT); + + /* Set the SYNC divider */ + sync_div = pcm->sclk_per_fs - 1; + + clkctl &= ~(S3C_PCM_CLKCTL_SYNCDIV_MASK + << S3C_PCM_CLKCTL_SYNCDIV_SHIFT); + clkctl |= ((sync_div & S3C_PCM_CLKCTL_SYNCDIV_MASK) + << S3C_PCM_CLKCTL_SYNCDIV_SHIFT); + + writel(clkctl, regs + S3C_PCM_CLKCTL); + + spin_unlock_irqrestore(&pcm->lock, flags); + + dev_dbg(pcm->dev, "PCMSOURCE_CLK-%lu SCLK=%ufs SCLK_DIV=%d SYNC_DIV=%d\n", + clk_get_rate(clk), pcm->sclk_per_fs, + sclk_div, sync_div); + + return 0; +} + +static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct s3c_pcm_info *pcm = to_info(cpu_dai); + void __iomem *regs = pcm->regs; + unsigned long flags; + int ret = 0; + u32 ctl; + + dev_dbg(pcm->dev, "Entered %s\n", __func__); + + spin_lock_irqsave(&pcm->lock, flags); + + ctl = readl(regs + S3C_PCM_CTL); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Nothing to do, NB_NF by default */ + break; + default: + dev_err(pcm->dev, "Unsupported clock inversion!\n"); + ret = -EINVAL; + goto exit; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* Nothing to do, Master by default */ + break; + default: + dev_err(pcm->dev, "Unsupported master/slave format!\n"); + ret = -EINVAL; + goto exit; + } + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + pcm->idleclk = 1; + break; + case SND_SOC_DAIFMT_GATED: + pcm->idleclk = 0; + break; + default: + dev_err(pcm->dev, "Invalid Clock gating request!\n"); + ret = -EINVAL; + goto exit; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + ctl |= S3C_PCM_CTL_TXMSB_AFTER_FSYNC; + ctl |= S3C_PCM_CTL_RXMSB_AFTER_FSYNC; + break; + case SND_SOC_DAIFMT_DSP_B: + ctl &= ~S3C_PCM_CTL_TXMSB_AFTER_FSYNC; + ctl &= ~S3C_PCM_CTL_RXMSB_AFTER_FSYNC; + break; + default: + dev_err(pcm->dev, "Unsupported data format!\n"); + ret = -EINVAL; + goto exit; + } + + writel(ctl, regs + S3C_PCM_CTL); + +exit: + spin_unlock_irqrestore(&pcm->lock, flags); + + return ret; +} + +static int s3c_pcm_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct s3c_pcm_info *pcm = to_info(cpu_dai); + + switch (div_id) { + case S3C_PCM_SCLK_PER_FS: + pcm->sclk_per_fs = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int s3c_pcm_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct s3c_pcm_info *pcm = to_info(cpu_dai); + void __iomem *regs = pcm->regs; + u32 clkctl = readl(regs + S3C_PCM_CLKCTL); + + switch (clk_id) { + case S3C_PCM_CLKSRC_PCLK: + clkctl |= S3C_PCM_CLKCTL_SERCLKSEL_PCLK; + break; + + case S3C_PCM_CLKSRC_MUX: + clkctl &= ~S3C_PCM_CLKCTL_SERCLKSEL_PCLK; + + if (clk_get_rate(pcm->cclk) != freq) + clk_set_rate(pcm->cclk, freq); + + break; + + default: + return -EINVAL; + } + + writel(clkctl, regs + S3C_PCM_CLKCTL); + + return 0; +} + +static struct snd_soc_dai_ops s3c_pcm_dai_ops = { + .set_sysclk = s3c_pcm_set_sysclk, + .set_clkdiv = s3c_pcm_set_clkdiv, + .trigger = s3c_pcm_trigger, + .hw_params = s3c_pcm_hw_params, + .set_fmt = s3c_pcm_set_fmt, +}; + +#define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 + +#define S3C_PCM_DECLARE(n) \ +{ \ + .name = "samsung-pcm", \ + .id = (n), \ + .symmetric_rates = 1, \ + .ops = &s3c_pcm_dai_ops, \ + .playback = { \ + .channels_min = 2, \ + .channels_max = 2, \ + .rates = S3C_PCM_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ + .capture = { \ + .channels_min = 2, \ + .channels_max = 2, \ + .rates = S3C_PCM_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ +} + +struct snd_soc_dai s3c_pcm_dai[] = { + S3C_PCM_DECLARE(0), + S3C_PCM_DECLARE(1), +}; +EXPORT_SYMBOL_GPL(s3c_pcm_dai); + +static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) +{ + struct s3c_pcm_info *pcm; + struct snd_soc_dai *dai; + struct resource *mem_res, *dmatx_res, *dmarx_res; + struct s3c_audio_pdata *pcm_pdata; + int ret; + + /* Check for valid device index */ + if ((pdev->id < 0) || pdev->id >= ARRAY_SIZE(s3c_pcm)) { + dev_err(&pdev->dev, "id %d out of range\n", pdev->id); + return -EINVAL; + } + + pcm_pdata = pdev->dev.platform_data; + + /* Check for availability of necessary resource */ + dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmatx_res) { + dev_err(&pdev->dev, "Unable to get PCM-TX dma resource\n"); + return -ENXIO; + } + + dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmarx_res) { + dev_err(&pdev->dev, "Unable to get PCM-RX dma resource\n"); + return -ENXIO; + } + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem_res) { + dev_err(&pdev->dev, "Unable to get register resource\n"); + return -ENXIO; + } + + if (pcm_pdata && pcm_pdata->cfg_gpio && pcm_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + return -EINVAL; + } + + pcm = &s3c_pcm[pdev->id]; + pcm->dev = &pdev->dev; + + spin_lock_init(&pcm->lock); + + dai = &s3c_pcm_dai[pdev->id]; + dai->dev = &pdev->dev; + + /* Default is 128fs */ + pcm->sclk_per_fs = 128; + + pcm->cclk = clk_get(&pdev->dev, "audio-bus"); + if (IS_ERR(pcm->cclk)) { + dev_err(&pdev->dev, "failed to get audio-bus\n"); + ret = PTR_ERR(pcm->cclk); + goto err1; + } + clk_enable(pcm->cclk); + + /* record our pcm structure for later use in the callbacks */ + dai->private_data = pcm; + + if (!request_mem_region(mem_res->start, + resource_size(mem_res), "samsung-pcm")) { + dev_err(&pdev->dev, "Unable to request register region\n"); + ret = -EBUSY; + goto err2; + } + + pcm->regs = ioremap(mem_res->start, 0x100); + if (pcm->regs == NULL) { + dev_err(&pdev->dev, "cannot ioremap registers\n"); + ret = -ENXIO; + goto err3; + } + + pcm->pclk = clk_get(&pdev->dev, "pcm"); + if (IS_ERR(pcm->pclk)) { + dev_err(&pdev->dev, "failed to get pcm_clock\n"); + ret = -ENOENT; + goto err4; + } + clk_enable(pcm->pclk); + + ret = snd_soc_register_dai(dai); + if (ret != 0) { + dev_err(&pdev->dev, "failed to get pcm_clock\n"); + goto err5; + } + + s3c_pcm_stereo_in[pdev->id].dma_addr = mem_res->start + + S3C_PCM_RXFIFO; + s3c_pcm_stereo_out[pdev->id].dma_addr = mem_res->start + + S3C_PCM_TXFIFO; + + s3c_pcm_stereo_in[pdev->id].channel = dmarx_res->start; + s3c_pcm_stereo_out[pdev->id].channel = dmatx_res->start; + + pcm->dma_capture = &s3c_pcm_stereo_in[pdev->id]; + pcm->dma_playback = &s3c_pcm_stereo_out[pdev->id]; + + return 0; + +err5: + clk_disable(pcm->pclk); + clk_put(pcm->pclk); +err4: + iounmap(pcm->regs); +err3: + release_mem_region(mem_res->start, resource_size(mem_res)); +err2: + clk_disable(pcm->cclk); + clk_put(pcm->cclk); +err1: + return ret; +} + +static __devexit int s3c_pcm_dev_remove(struct platform_device *pdev) +{ + struct s3c_pcm_info *pcm = &s3c_pcm[pdev->id]; + struct resource *mem_res; + + iounmap(pcm->regs); + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(mem_res->start, resource_size(mem_res)); + + clk_disable(pcm->cclk); + clk_disable(pcm->pclk); + clk_put(pcm->pclk); + clk_put(pcm->cclk); + + return 0; +} + +static struct platform_driver s3c_pcm_driver = { + .probe = s3c_pcm_dev_probe, + .remove = s3c_pcm_dev_remove, + .driver = { + .name = "samsung-pcm", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c_pcm_init(void) +{ + return platform_driver_register(&s3c_pcm_driver); +} +module_init(s3c_pcm_init); + +static void __exit s3c_pcm_exit(void) +{ + platform_driver_unregister(&s3c_pcm_driver); +} +module_exit(s3c_pcm_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_DESCRIPTION("S3C PCM Controller Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c-pcm.h b/sound/soc/samsung/s3c-pcm.h new file mode 100644 index 0000000..69ff997 --- /dev/null +++ b/sound/soc/samsung/s3c-pcm.h @@ -0,0 +1,123 @@ +/* sound/soc/s3c24xx/s3c-pcm.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __S3C_PCM_H +#define __S3C_PCM_H __FILE__ + +/*Register Offsets */ +#define S3C_PCM_CTL (0x00) +#define S3C_PCM_CLKCTL (0x04) +#define S3C_PCM_TXFIFO (0x08) +#define S3C_PCM_RXFIFO (0x0C) +#define S3C_PCM_IRQCTL (0x10) +#define S3C_PCM_IRQSTAT (0x14) +#define S3C_PCM_FIFOSTAT (0x18) +#define S3C_PCM_CLRINT (0x20) + +/* PCM_CTL Bit-Fields */ +#define S3C_PCM_CTL_TXDIPSTICK_MASK (0x3f) +#define S3C_PCM_CTL_TXDIPSTICK_SHIFT (13) +#define S3C_PCM_CTL_RXDIPSTICK_MSK (0x3f<<7) +#define S3C_PCM_CTL_TXDMA_EN (0x1<<6) +#define S3C_PCM_CTL_RXDMA_EN (0x1<<5) +#define S3C_PCM_CTL_TXMSB_AFTER_FSYNC (0x1<<4) +#define S3C_PCM_CTL_RXMSB_AFTER_FSYNC (0x1<<3) +#define S3C_PCM_CTL_TXFIFO_EN (0x1<<2) +#define S3C_PCM_CTL_RXFIFO_EN (0x1<<1) +#define S3C_PCM_CTL_ENABLE (0x1<<0) + +/* PCM_CLKCTL Bit-Fields */ +#define S3C_PCM_CLKCTL_SERCLK_EN (0x1<<19) +#define S3C_PCM_CLKCTL_SERCLKSEL_PCLK (0x1<<18) +#define S3C_PCM_CLKCTL_SCLKDIV_MASK (0x1ff) +#define S3C_PCM_CLKCTL_SYNCDIV_MASK (0x1ff) +#define S3C_PCM_CLKCTL_SCLKDIV_SHIFT (9) +#define S3C_PCM_CLKCTL_SYNCDIV_SHIFT (0) + +/* PCM_TXFIFO Bit-Fields */ +#define S3C_PCM_TXFIFO_DVALID (0x1<<16) +#define S3C_PCM_TXFIFO_DATA_MSK (0xffff<<0) + +/* PCM_RXFIFO Bit-Fields */ +#define S3C_PCM_RXFIFO_DVALID (0x1<<16) +#define S3C_PCM_RXFIFO_DATA_MSK (0xffff<<0) + +/* PCM_IRQCTL Bit-Fields */ +#define S3C_PCM_IRQCTL_IRQEN (0x1<<14) +#define S3C_PCM_IRQCTL_WRDEN (0x1<<12) +#define S3C_PCM_IRQCTL_TXEMPTYEN (0x1<<11) +#define S3C_PCM_IRQCTL_TXALMSTEMPTYEN (0x1<<10) +#define S3C_PCM_IRQCTL_TXFULLEN (0x1<<9) +#define S3C_PCM_IRQCTL_TXALMSTFULLEN (0x1<<8) +#define S3C_PCM_IRQCTL_TXSTARVEN (0x1<<7) +#define S3C_PCM_IRQCTL_TXERROVRFLEN (0x1<<6) +#define S3C_PCM_IRQCTL_RXEMPTEN (0x1<<5) +#define S3C_PCM_IRQCTL_RXALMSTEMPTEN (0x1<<4) +#define S3C_PCM_IRQCTL_RXFULLEN (0x1<<3) +#define S3C_PCM_IRQCTL_RXALMSTFULLEN (0x1<<2) +#define S3C_PCM_IRQCTL_RXSTARVEN (0x1<<1) +#define S3C_PCM_IRQCTL_RXERROVRFLEN (0x1<<0) + +/* PCM_IRQSTAT Bit-Fields */ +#define S3C_PCM_IRQSTAT_IRQPND (0x1<<13) +#define S3C_PCM_IRQSTAT_WRD_XFER (0x1<<12) +#define S3C_PCM_IRQSTAT_TXEMPTY (0x1<<11) +#define S3C_PCM_IRQSTAT_TXALMSTEMPTY (0x1<<10) +#define S3C_PCM_IRQSTAT_TXFULL (0x1<<9) +#define S3C_PCM_IRQSTAT_TXALMSTFULL (0x1<<8) +#define S3C_PCM_IRQSTAT_TXSTARV (0x1<<7) +#define S3C_PCM_IRQSTAT_TXERROVRFL (0x1<<6) +#define S3C_PCM_IRQSTAT_RXEMPT (0x1<<5) +#define S3C_PCM_IRQSTAT_RXALMSTEMPT (0x1<<4) +#define S3C_PCM_IRQSTAT_RXFULL (0x1<<3) +#define S3C_PCM_IRQSTAT_RXALMSTFULL (0x1<<2) +#define S3C_PCM_IRQSTAT_RXSTARV (0x1<<1) +#define S3C_PCM_IRQSTAT_RXERROVRFL (0x1<<0) + +/* PCM_FIFOSTAT Bit-Fields */ +#define S3C_PCM_FIFOSTAT_TXCNT_MSK (0x3f<<14) +#define S3C_PCM_FIFOSTAT_TXFIFOEMPTY (0x1<<13) +#define S3C_PCM_FIFOSTAT_TXFIFOALMSTEMPTY (0x1<<12) +#define S3C_PCM_FIFOSTAT_TXFIFOFULL (0x1<<11) +#define S3C_PCM_FIFOSTAT_TXFIFOALMSTFULL (0x1<<10) +#define S3C_PCM_FIFOSTAT_RXCNT_MSK (0x3f<<4) +#define S3C_PCM_FIFOSTAT_RXFIFOEMPTY (0x1<<3) +#define S3C_PCM_FIFOSTAT_RXFIFOALMSTEMPTY (0x1<<2) +#define S3C_PCM_FIFOSTAT_RXFIFOFULL (0x1<<1) +#define S3C_PCM_FIFOSTAT_RXFIFOALMSTFULL (0x1<<0) + +#define S3C_PCM_CLKSRC_PCLK 0 +#define S3C_PCM_CLKSRC_MUX 1 + +#define S3C_PCM_SCLK_PER_FS 0 + +/** + * struct s3c_pcm_info - S3C PCM Controller information + * @dev: The parent device passed to use from the probe. + * @regs: The pointer to the device register block. + * @dma_playback: DMA information for playback channel. + * @dma_capture: DMA information for capture channel. + */ +struct s3c_pcm_info { + spinlock_t lock; + struct device *dev; + void __iomem *regs; + + unsigned int sclk_per_fs; + + /* Whether to keep PCMSCLK enabled even when idle(no active xfer) */ + unsigned int idleclk; + + struct clk *pclk; + struct clk *cclk; + + struct s3c_dma_params *dma_playback; + struct s3c_dma_params *dma_capture; +}; + +#endif /* __S3C_PCM_H */ diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c new file mode 100644 index 0000000..709adef --- /dev/null +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -0,0 +1,190 @@ +/* sound/soc/s3c24xx/s3c2412-i2s.c + * + * ALSA Soc Audio Layer - S3C2412 I2S driver + * + * Copyright (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory(a)wolfsonmicro.com + * linux(a)wolfsonmicro.com + * + * Copyright (c) 2007, 2004-2005 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <linux/kernel.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <mach/hardware.h> + +#include <mach/regs-gpio.h> +#include <mach/dma.h> + +#include "s3c-dma.h" +#include "regs-i2s-v2.h" +#include "s3c2412-i2s.h" + +#define S3C2412_I2S_DEBUG 0 + +static struct s3c2410_dma_client s3c2412_dma_client_out = { + .name = "I2S PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c2412_dma_client_in = { + .name = "I2S PCM Stereo in" +}; + +static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = { + .client = &s3c2412_dma_client_out, + .channel = DMACH_I2S_OUT, + .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, + .dma_size = 4, +}; + +static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { + .client = &s3c2412_dma_client_in, + .channel = DMACH_I2S_IN, + .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, + .dma_size = 4, +}; + +static struct s3c_i2sv2_info s3c2412_i2s; + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static int s3c2412_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + int ret; + + pr_debug("Entered %s\n", __func__); + + ret = s3c_i2sv2_probe(pdev, dai, &s3c2412_i2s, S3C2410_PA_IIS); + if (ret) + return ret; + + s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in; + s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; + + s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk"); + if (s3c2412_i2s.iis_cclk == NULL) { + pr_err("failed to get i2sclk clock\n"); + iounmap(s3c2412_i2s.regs); + return -ENODEV; + } + + /* Set MPLL as the source for IIS CLK */ + + clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); + clk_enable(s3c2412_i2s.iis_cclk); + + s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk; + + /* Configure the I2S pins in correct mode */ + s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); + s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); + s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); + + return 0; +} + +static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + struct s3c_dma_params *dma_data; + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = i2s->dma_playback; + else + dma_data = i2s->dma_capture; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + + iismod = readl(i2s->regs + S3C2412_IISMOD); + pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod |= S3C2412_IISMOD_8BIT; + break; + case SNDRV_PCM_FORMAT_S16_LE: + iismod &= ~S3C2412_IISMOD_8BIT; + break; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); + + return 0; +} + +#define S3C2412_I2S_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = { + .hw_params = s3c2412_i2s_hw_params, +}; + +struct snd_soc_dai s3c2412_i2s_dai = { + .name = "s3c2412-i2s", + .id = 0, + .probe = s3c2412_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C2412_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C2412_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &s3c2412_i2s_dai_ops, +}; +EXPORT_SYMBOL_GPL(s3c2412_i2s_dai); + +static int __init s3c2412_i2s_init(void) +{ + return s3c_i2sv2_register_dai(&s3c2412_i2s_dai); +} +module_init(s3c2412_i2s_init); + +static void __exit s3c2412_i2s_exit(void) +{ + snd_soc_unregister_dai(&s3c2412_i2s_dai); +} +module_exit(s3c2412_i2s_exit); + +/* Module information */ +MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("S3C2412 I2S SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c2412-i2s.h b/sound/soc/samsung/s3c2412-i2s.h new file mode 100644 index 0000000..0b5686b --- /dev/null +++ b/sound/soc/samsung/s3c2412-i2s.h @@ -0,0 +1,29 @@ +/* sound/soc/s3c24xx/s3c2412-i2s.c + * + * ALSA Soc Audio Layer - S3C2412 I2S driver + * + * Copyright (c) 2007 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. +*/ + +#ifndef __SND_SOC_S3C24XX_S3C2412_I2S_H +#define __SND_SOC_S3C24XX_S3C2412_I2S_H __FILE__ + +#include "s3c-i2s-v2.h" + +#define S3C2412_DIV_BCLK S3C_I2SV2_DIV_BCLK +#define S3C2412_DIV_RCLK S3C_I2SV2_DIV_RCLK +#define S3C2412_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER + +#define S3C2412_CLKSRC_PCLK S3C_I2SV2_CLKSRC_PCLK +#define S3C2412_CLKSRC_I2SCLK S3C_I2SV2_CLKSRC_AUDIOBUS + +extern struct snd_soc_dai s3c2412_i2s_dai; + +#endif /* __SND_SOC_S3C24XX_S3C2412_I2S_H */ diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c new file mode 100644 index 0000000..c3ac890 --- /dev/null +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -0,0 +1,503 @@ +/* + * s3c24xx-i2s.c -- ALSA Soc Audio Layer + * + * (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * Copyright 2004-2005 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/jiffies.h> +#include <linux/io.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/hardware.h> +#include <mach/regs-gpio.h> +#include <mach/regs-clock.h> + +#include <asm/dma.h> +#include <mach/dma.h> + +#include <plat/regs-iis.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" + +static struct s3c2410_dma_client s3c24xx_dma_client_out = { + .name = "I2S PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c24xx_dma_client_in = { + .name = "I2S PCM Stereo in" +}; + +static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = { + .client = &s3c24xx_dma_client_out, + .channel = DMACH_I2S_OUT, + .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .dma_size = 2, +}; + +static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = { + .client = &s3c24xx_dma_client_in, + .channel = DMACH_I2S_IN, + .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .dma_size = 2, +}; + +struct s3c24xx_i2s_info { + void __iomem *regs; + struct clk *iis_clk; + u32 iiscon; + u32 iismod; + u32 iisfcon; + u32 iispsr; +}; +static struct s3c24xx_i2s_info s3c24xx_i2s; + +static void s3c24xx_snd_txctrl(int on) +{ + u32 iisfcon; + u32 iiscon; + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); + iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); + iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + + pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); + + if (on) { + iisfcon |= S3C2410_IISFCON_TXDMA | S3C2410_IISFCON_TXENABLE; + iiscon |= S3C2410_IISCON_TXDMAEN | S3C2410_IISCON_IISEN; + iiscon &= ~S3C2410_IISCON_TXIDLE; + iismod |= S3C2410_IISMOD_TXMODE; + + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); + writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); + } else { + /* note, we have to disable the FIFOs otherwise bad things + * seem to happen when the DMA stops. According to the + * Samsung supplied kernel, this should allow the DMA + * engine and FIFOs to reset. If this isn't allowed, the + * DMA engine will simply freeze randomly. + */ + + iisfcon &= ~S3C2410_IISFCON_TXENABLE; + iisfcon &= ~S3C2410_IISFCON_TXDMA; + iiscon |= S3C2410_IISCON_TXIDLE; + iiscon &= ~S3C2410_IISCON_TXDMAEN; + iismod &= ~S3C2410_IISMOD_TXMODE; + + writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); + writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + } + + pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); +} + +static void s3c24xx_snd_rxctrl(int on) +{ + u32 iisfcon; + u32 iiscon; + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); + iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); + iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + + pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); + + if (on) { + iisfcon |= S3C2410_IISFCON_RXDMA | S3C2410_IISFCON_RXENABLE; + iiscon |= S3C2410_IISCON_RXDMAEN | S3C2410_IISCON_IISEN; + iiscon &= ~S3C2410_IISCON_RXIDLE; + iismod |= S3C2410_IISMOD_RXMODE; + + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); + writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); + } else { + /* note, we have to disable the FIFOs otherwise bad things + * seem to happen when the DMA stops. According to the + * Samsung supplied kernel, this should allow the DMA + * engine and FIFOs to reset. If this isn't allowed, the + * DMA engine will simply freeze randomly. + */ + + iisfcon &= ~S3C2410_IISFCON_RXENABLE; + iisfcon &= ~S3C2410_IISFCON_RXDMA; + iiscon |= S3C2410_IISCON_RXIDLE; + iiscon &= ~S3C2410_IISCON_RXDMAEN; + iismod &= ~S3C2410_IISMOD_RXMODE; + + writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); + writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + } + + pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); +} + +/* + * Wait for the LR signal to allow synchronisation to the L/R clock + * from the codec. May only be needed for slave mode. + */ +static int s3c24xx_snd_lrsync(void) +{ + u32 iiscon; + int timeout = 50; /* 5ms */ + + pr_debug("Entered %s\n", __func__); + + while (1) { + iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); + if (iiscon & S3C2410_IISCON_LRINDEX) + break; + + if (!timeout--) + return -ETIMEDOUT; + udelay(100); + } + + return 0; +} + +/* + * Check whether CPU is the master or slave + */ +static inline int s3c24xx_snd_is_clkmaster(void) +{ + pr_debug("Entered %s\n", __func__); + + return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1; +} + +/* + * Set S3C24xx I2S DAI format + */ +static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + pr_debug("hw_params r: IISMOD: %x \n", iismod); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iismod |= S3C2410_IISMOD_SLAVE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iismod &= ~S3C2410_IISMOD_SLAVE; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + iismod |= S3C2410_IISMOD_MSB; + break; + case SND_SOC_DAIFMT_I2S: + iismod &= ~S3C2410_IISMOD_MSB; + break; + default: + return -EINVAL; + } + + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + pr_debug("hw_params w: IISMOD: %x \n", iismod); + return 0; +} + +static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_dma_params *dma_data; + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = &s3c24xx_i2s_pcm_stereo_out; + else + dma_data = &s3c24xx_i2s_pcm_stereo_in; + + snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_data); + + /* Working copies of register */ + iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + pr_debug("hw_params r: IISMOD: %x\n", iismod); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod &= ~S3C2410_IISMOD_16BIT; + dma_data->dma_size = 1; + break; + case SNDRV_PCM_FORMAT_S16_LE: + iismod |= S3C2410_IISMOD_16BIT; + dma_data->dma_size = 2; + break; + default: + return -EINVAL; + } + + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + pr_debug("hw_params w: IISMOD: %x\n", iismod); + return 0; +} + +static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + + pr_debug("Entered %s\n", __func__); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!s3c24xx_snd_is_clkmaster()) { + ret = s3c24xx_snd_lrsync(); + if (ret) + goto exit_err; + } + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + s3c24xx_snd_rxctrl(1); + else + s3c24xx_snd_txctrl(1); + + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + s3c24xx_snd_rxctrl(0); + else + s3c24xx_snd_txctrl(0); + break; + default: + ret = -EINVAL; + break; + } + +exit_err: + return ret; +} + +/* + * Set S3C24xx Clock source + */ +static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + + pr_debug("Entered %s\n", __func__); + + iismod &= ~S3C2440_IISMOD_MPLL; + + switch (clk_id) { + case S3C24XX_CLKSRC_PCLK: + break; + case S3C24XX_CLKSRC_MPLL: + iismod |= S3C2440_IISMOD_MPLL; + break; + default: + return -EINVAL; + } + + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + return 0; +} + +/* + * Set S3C24xx Clock dividers + */ +static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + u32 reg; + + pr_debug("Entered %s\n", __func__); + + switch (div_id) { + case S3C24XX_DIV_BCLK: + reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~S3C2410_IISMOD_FS_MASK; + writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); + break; + case S3C24XX_DIV_MCLK: + reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~(S3C2410_IISMOD_384FS); + writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); + break; + case S3C24XX_DIV_PRESCALER: + writel(div, s3c24xx_i2s.regs + S3C2410_IISPSR); + reg = readl(s3c24xx_i2s.regs + S3C2410_IISCON); + writel(reg | S3C2410_IISCON_PSCEN, s3c24xx_i2s.regs + S3C2410_IISCON); + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * To avoid duplicating clock code, allow machine driver to + * get the clockrate from here. + */ +u32 s3c24xx_i2s_get_clockrate(void) +{ + return clk_get_rate(s3c24xx_i2s.iis_clk); +} +EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); + +static int s3c24xx_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + pr_debug("Entered %s\n", __func__); + + s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); + if (s3c24xx_i2s.regs == NULL) + return -ENXIO; + + s3c24xx_i2s.iis_clk = clk_get(&pdev->dev, "iis"); + if (s3c24xx_i2s.iis_clk == NULL) { + pr_err("failed to get iis_clock\n"); + iounmap(s3c24xx_i2s.regs); + return -ENODEV; + } + clk_enable(s3c24xx_i2s.iis_clk); + + /* Configure the I2S pins in correct mode */ + s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); + s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); + s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); + + writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON); + + s3c24xx_snd_txctrl(0); + s3c24xx_snd_rxctrl(0); + + return 0; +} + +#ifdef CONFIG_PM +static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai) +{ + pr_debug("Entered %s\n", __func__); + + s3c24xx_i2s.iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); + s3c24xx_i2s.iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + s3c24xx_i2s.iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); + s3c24xx_i2s.iispsr = readl(s3c24xx_i2s.regs + S3C2410_IISPSR); + + clk_disable(s3c24xx_i2s.iis_clk); + + return 0; +} + +static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai) +{ + pr_debug("Entered %s\n", __func__); + clk_enable(s3c24xx_i2s.iis_clk); + + writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); + writel(s3c24xx_i2s.iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + writel(s3c24xx_i2s.iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); + writel(s3c24xx_i2s.iispsr, s3c24xx_i2s.regs + S3C2410_IISPSR); + + return 0; +} +#else +#define s3c24xx_i2s_suspend NULL +#define s3c24xx_i2s_resume NULL +#endif + + +#define S3C24XX_I2S_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +static struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = { + .trigger = s3c24xx_i2s_trigger, + .hw_params = s3c24xx_i2s_hw_params, + .set_fmt = s3c24xx_i2s_set_fmt, + .set_clkdiv = s3c24xx_i2s_set_clkdiv, + .set_sysclk = s3c24xx_i2s_set_sysclk, +}; + +struct snd_soc_dai s3c24xx_i2s_dai = { + .name = "s3c24xx-i2s", + .id = 0, + .probe = s3c24xx_i2s_probe, + .suspend = s3c24xx_i2s_suspend, + .resume = s3c24xx_i2s_resume, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C24XX_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C24XX_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &s3c24xx_i2s_dai_ops, +}; +EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai); + +static int __init s3c24xx_i2s_init(void) +{ + return snd_soc_register_dai(&s3c24xx_i2s_dai); +} +module_init(s3c24xx_i2s_init); + +static void __exit s3c24xx_i2s_exit(void) +{ + snd_soc_unregister_dai(&s3c24xx_i2s_dai); +} +module_exit(s3c24xx_i2s_exit); + +/* Module information */ +MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("s3c24xx I2S SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx-i2s.h b/sound/soc/samsung/s3c24xx-i2s.h new file mode 100644 index 0000000..726d91c --- /dev/null +++ b/sound/soc/samsung/s3c24xx-i2s.h @@ -0,0 +1,37 @@ +/* + * s3c24xx-i2s.c -- ALSA Soc Audio Layer + * + * Copyright 2005 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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 + * 10th Nov 2006 Initial version. + */ + +#ifndef S3C24XXI2S_H_ +#define S3C24XXI2S_H_ + +/* clock sources */ +#define S3C24XX_CLKSRC_PCLK 0 +#define S3C24XX_CLKSRC_MPLL 1 + +/* Clock dividers */ +#define S3C24XX_DIV_MCLK 0 +#define S3C24XX_DIV_BCLK 1 +#define S3C24XX_DIV_PRESCALER 2 + +/* prescaler */ +#define S3C24XX_PRESCALE(a,b) \ + (((a - 1) << S3C2410_IISPSR_INTSHIFT) | ((b - 1) << S3C2410_IISPSR_EXTSHFIT)) + +u32 s3c24xx_i2s_get_clockrate(void); + +extern struct snd_soc_dai s3c24xx_i2s_dai; + +#endif /*S3C24XXI2S_H_*/ diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c new file mode 100644 index 0000000..4984754 --- /dev/null +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -0,0 +1,394 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec.c + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <linux/i2c.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +static struct s3c24xx_audio_simtec_pdata *pdata; +static struct clk *xtal_clk; + +static int spk_gain; +static int spk_unmute; + +/** + * speaker_gain_get - read the speaker gain setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be updated. + * + * Read the value for the AMP gain control. + */ +static int speaker_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = spk_gain; + return 0; +} + +/** + * speaker_gain_set - set the value of the speaker amp gain + * @value: The value to write. + */ +static void speaker_gain_set(int value) +{ + gpio_set_value_cansleep(pdata->amp_gain[0], value & 1); + gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1); +} + +/** + * speaker_gain_put - set the speaker gain setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be set. + * + * Set the value of the speaker gain from the specified + * @ucontrol setting. + * + * Note, if the speaker amp is muted, then we do not set a gain value + * as at-least one of the ICs that is fitted will try and power up even + * if the main control is set to off. + */ +static int speaker_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int value = ucontrol->value.integer.value[0]; + + spk_gain = value; + + if (!spk_unmute) + speaker_gain_set(value); + + return 0; +} + +static const struct snd_kcontrol_new amp_gain_controls[] = { + SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0, + speaker_gain_get, speaker_gain_put), +}; + +/** + * spk_unmute_state - set the unmute state of the speaker + * @to: zero to unmute, non-zero to ununmute. + */ +static void spk_unmute_state(int to) +{ + pr_debug("%s: to=%d\n", __func__, to); + + spk_unmute = to; + gpio_set_value(pdata->amp_gpio, to); + + /* if we're umuting, also re-set the gain */ + if (to && pdata->amp_gain[0] > 0) + speaker_gain_set(spk_gain); +} + +/** + * speaker_unmute_get - read the speaker unmute setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be updated. + * + * Read the value for the AMP gain control. + */ +static int speaker_unmute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = spk_unmute; + return 0; +} + +/** + * speaker_unmute_put - set the speaker unmute setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be set. + * + * Set the value of the speaker gain from the specified + * @ucontrol setting. + */ +static int speaker_unmute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + spk_unmute_state(ucontrol->value.integer.value[0]); + return 0; +} + +/* This is added as a manual control as the speaker amps create clicks + * when their power state is changed, which are far more noticeable than + * anything produced by the CODEC itself. + */ +static const struct snd_kcontrol_new amp_unmute_controls[] = { + SOC_SINGLE_EXT("Speaker Switch", 0, 0, 1, 0, + speaker_unmute_get, speaker_unmute_put), +}; + +void simtec_audio_init(struct snd_soc_codec *codec) +{ + if (pdata->amp_gpio > 0) { + pr_debug("%s: adding amp routes\n", __func__); + + snd_soc_add_controls(codec, amp_unmute_controls, + ARRAY_SIZE(amp_unmute_controls)); + } + + if (pdata->amp_gain[0] > 0) { + pr_debug("%s: adding amp controls\n", __func__); + snd_soc_add_controls(codec, amp_gain_controls, + ARRAY_SIZE(amp_gain_controls)); + } +} +EXPORT_SYMBOL_GPL(simtec_audio_init); + +#define CODEC_CLOCK 12000000 + +/** + * simtec_hw_params - update hardware parameters + * @substream: The audio substream instance. + * @params: The parameters requested. + * + * Update the codec data routing and configuration settings + * from the supplied data. + */ +static int simtec_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; + + /* Set the CODEC as the bus clock master, I2S */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set cpu dai format\n", __func__); + return ret; + } + + /* Set the CODEC as the bus clock master */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set codec dai format\n", __func__); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + CODEC_CLOCK, SND_SOC_CLOCK_IN); + if (ret) { + pr_err( "%s: failed setting codec sysclk\n", __func__); + return ret; + } + + if (pdata->use_mpllin) { + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL, + 0, SND_SOC_CLOCK_OUT); + + if (ret) { + pr_err("%s: failed to set MPLLin as clksrc\n", + __func__); + return ret; + } + } + + if (pdata->output_cdclk) { + int cdclk_scale; + + cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK; + cdclk_scale--; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + cdclk_scale); + } + + return 0; +} + +static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd) +{ + /* call any board supplied startup code, this currently only + * covers the bast/vr1000 which have a CPLD in the way of the + * LRCLK */ + if (pd->startup) + pd->startup(); + + return 0; +} + +static struct snd_soc_ops simtec_snd_ops = { + .hw_params = simtec_hw_params, +}; + +/** + * attach_gpio_amp - get and configure the necessary gpios + * @dev: The device we're probing. + * @pd: The platform data supplied by the board. + * + * If there is a GPIO based amplifier attached to the board, claim + * the necessary GPIO lines for it, and set default values. + */ +static int attach_gpio_amp(struct device *dev, + struct s3c24xx_audio_simtec_pdata *pd) +{ + int ret; + + /* attach gpio amp gain (if any) */ + if (pdata->amp_gain[0] > 0) { + ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain0"); + if (ret) { + dev_err(dev, "cannot get amp gpio gain0\n"); + return ret; + } + + ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain1"); + if (ret) { + dev_err(dev, "cannot get amp gpio gain1\n"); + gpio_free(pdata->amp_gain[0]); + return ret; + } + + gpio_direction_output(pd->amp_gain[0], 0); + gpio_direction_output(pd->amp_gain[1], 0); + } + + /* note, currently we assume GPA0 isn't valid amp */ + if (pdata->amp_gpio > 0) { + ret = gpio_request(pd->amp_gpio, "gpio-amp"); + if (ret) { + dev_err(dev, "cannot get amp gpio %d (%d)\n", + pd->amp_gpio, ret); + goto err_amp; + } + + /* set the amp off at startup */ + spk_unmute_state(0); + } + + return 0; + +err_amp: + if (pd->amp_gain[0] > 0) { + gpio_free(pd->amp_gain[0]); + gpio_free(pd->amp_gain[1]); + } + + return ret; +} + +static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) +{ + if (pd->amp_gain[0] > 0) { + gpio_free(pd->amp_gain[0]); + gpio_free(pd->amp_gain[1]); + } + + if (pd->amp_gpio > 0) + gpio_free(pd->amp_gpio); +} + +#ifdef CONFIG_PM +int simtec_audio_resume(struct device *dev) +{ + simtec_call_startup(pdata); + return 0; +} + +const struct dev_pm_ops simtec_audio_pmops = { + .resume = simtec_audio_resume, +}; +EXPORT_SYMBOL_GPL(simtec_audio_pmops); +#endif + +int __devinit simtec_audio_core_probe(struct platform_device *pdev, + struct snd_soc_device *socdev) +{ + struct platform_device *snd_dev; + int ret; + + socdev->card->dai_link->ops = &simtec_snd_ops; + + pdata = pdev->dev.platform_data; + if (!pdata) { + dev_err(&pdev->dev, "no platform data supplied\n"); + return -EINVAL; + } + + simtec_call_startup(pdata); + + xtal_clk = clk_get(&pdev->dev, "xtal"); + if (IS_ERR(xtal_clk)) { + dev_err(&pdev->dev, "could not get clkout0\n"); + return -EINVAL; + } + + dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk)); + + ret = attach_gpio_amp(&pdev->dev, pdata); + if (ret) + goto err_clk; + + snd_dev = platform_device_alloc("soc-audio", -1); + if (!snd_dev) { + dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n"); + ret = -ENOMEM; + goto err_gpio; + } + + platform_set_drvdata(snd_dev, socdev); + socdev->dev = &snd_dev->dev; + + ret = platform_device_add(snd_dev); + if (ret) { + dev_err(&pdev->dev, "failed to add soc-audio dev\n"); + goto err_pdev; + } + + platform_set_drvdata(pdev, snd_dev); + return 0; + +err_pdev: + platform_device_put(snd_dev); + +err_gpio: + detach_gpio_amp(pdata); + +err_clk: + clk_put(xtal_clk); + return ret; +} +EXPORT_SYMBOL_GPL(simtec_audio_core_probe); + +int __devexit simtec_audio_remove(struct platform_device *pdev) +{ + struct platform_device *snd_dev = platform_get_drvdata(pdev); + + platform_device_unregister(snd_dev); + + detach_gpio_amp(pdata); + clk_put(xtal_clk); + return 0; +} +EXPORT_SYMBOL_GPL(simtec_audio_remove); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio common support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx_simtec.h b/sound/soc/samsung/s3c24xx_simtec.h new file mode 100644 index 0000000..e18faee --- /dev/null +++ b/sound/soc/samsung/s3c24xx_simtec.h @@ -0,0 +1,22 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec.h + * + * Copyright 2009 Simtec Electronics + * + * 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. +*/ + +extern void simtec_audio_init(struct snd_soc_codec *codec); + +extern int simtec_audio_core_probe(struct platform_device *pdev, + struct snd_soc_device *socdev); + +extern int simtec_audio_remove(struct platform_device *pdev); + +#ifdef CONFIG_PM +extern const struct dev_pm_ops simtec_audio_pmops; +#define simtec_audio_pm &simtec_audio_pmops +#else +#define simtec_audio_pm NULL +#endif diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c new file mode 100644 index 0000000..bdf8951 --- /dev/null +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -0,0 +1,153 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec_hermes.c + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include <linux/module.h> +#include <linux/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 <plat/audio-simtec.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +#include "../codecs/tlv320aic3x.h" + +static const struct snd_soc_dapm_widget dapm_widgets[] = { + SND_SOC_DAPM_LINE("GSM Out", NULL), + SND_SOC_DAPM_LINE("GSM In", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_LINE("ZV", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), +}; + +static const struct snd_soc_dapm_route base_map[] = { + /* Headphone connected to HP{L,R}OUT and HP{L,R}COM */ + + { "Headphone Jack", NULL, "HPLOUT" }, + { "Headphone Jack", NULL, "HPLCOM" }, + { "Headphone Jack", NULL, "HPROUT" }, + { "Headphone Jack", NULL, "HPRCOM" }, + + /* ZV connected to Line1 */ + + { "LINE1L", NULL, "ZV" }, + { "LINE1R", NULL, "ZV" }, + + /* Line In connected to Line2 */ + + { "LINE2L", NULL, "Line In" }, + { "LINE2R", NULL, "Line In" }, + + /* Microphone connected to MIC3R and MIC_BIAS */ + + { "MIC3L", NULL, "Mic Jack" }, + + /* GSM connected to MONO_LOUT and MIC3L (in) */ + + { "GSM Out", NULL, "MONO_LOUT" }, + { "MIC3L", NULL, "GSM In" }, + + /* Speaker is connected to LINEOUT{LN,LP,RN,RP}, however we are + * not using the DAPM to power it up and down as there it makes + * a click when powering up. */ +}; + +/** + * simtec_hermes_init - initialise and add controls + * @codec; The codec instance to attach to. + * + * Attach our controls and configure the necessary codec + * mappings for our sound card instance. +*/ +static int simtec_hermes_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dapm_widgets, + ARRAY_SIZE(dapm_widgets)); + + snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + simtec_audio_init(codec); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct aic3x_setup_data codec_setup = { +}; + +static struct snd_soc_dai_link simtec_dai_aic33 = { + .name = "tlv320aic33", + .stream_name = "TLV320AIC33", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &aic3x_dai, + .init = simtec_hermes_init, +}; + +/* simtec audio machine driver */ +static struct snd_soc_card snd_soc_machine_simtec_aic33 = { + .name = "Simtec-Hermes", + .platform = &s3c24xx_soc_platform, + .dai_link = &simtec_dai_aic33, + .num_links = 1, +}; + +/* simtec audio subsystem */ +static struct snd_soc_device simtec_snd_devdata_aic33 = { + .card = &snd_soc_machine_simtec_aic33, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &codec_setup, +}; + +static int __devinit simtec_audio_hermes_probe(struct platform_device *pd) +{ + dev_info(&pd->dev, "probing....\n"); + return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic33); +} + +static struct platform_driver simtec_audio_hermes_platdrv = { + .driver = { + .owner = THIS_MODULE, + .name = "s3c24xx-simtec-hermes-snd", + .pm = simtec_audio_pm, + }, + .probe = simtec_audio_hermes_probe, + .remove = __devexit_p(simtec_audio_remove), +}; + +MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd"); + +static int __init simtec_hermes_modinit(void) +{ + return platform_driver_register(&simtec_audio_hermes_platdrv); +} + +static void __exit simtec_hermes_modexit(void) +{ + platform_driver_unregister(&simtec_audio_hermes_platdrv); +} + +module_init(simtec_hermes_modinit); +module_exit(simtec_hermes_modexit); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c new file mode 100644 index 0000000..185c0ac --- /dev/null +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -0,0 +1,137 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include <linux/module.h> +#include <linux/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 <plat/audio-simtec.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +#include "../codecs/tlv320aic23.h" + +/* supported machines: + * + * Machine Connections AMP + * ------- ----------- --- + * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired) + * VR1000 HPOUT, LIN None + * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) + * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) + * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R) + */ + +static const struct snd_soc_dapm_widget dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route base_map[] = { + { "Headphone Jack", NULL, "LHPOUT"}, + { "Headphone Jack", NULL, "RHPOUT"}, + + { "Line Out", NULL, "LOUT" }, + { "Line Out", NULL, "ROUT" }, + + { "LLINEIN", NULL, "Line In"}, + { "RLINEIN", NULL, "Line In"}, + + { "MICIN", NULL, "Mic Jack"}, +}; + +/** + * simtec_tlv320aic23_init - initialise and add controls + * @codec; The codec instance to attach to. + * + * Attach our controls and configure the necessary codec + * mappings for our sound card instance. +*/ +static int simtec_tlv320aic23_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dapm_widgets, + ARRAY_SIZE(dapm_widgets)); + + snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + simtec_audio_init(codec); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link simtec_dai_aic23 = { + .name = "tlv320aic23", + .stream_name = "TLV320AIC23", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &tlv320aic23_dai, + .init = simtec_tlv320aic23_init, +}; + +/* simtec audio machine driver */ +static struct snd_soc_card snd_soc_machine_simtec_aic23 = { + .name = "Simtec", + .platform = &s3c24xx_soc_platform, + .dai_link = &simtec_dai_aic23, + .num_links = 1, +}; + +/* simtec audio subsystem */ +static struct snd_soc_device simtec_snd_devdata_aic23 = { + .card = &snd_soc_machine_simtec_aic23, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd) +{ + return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic23); +} + +static struct platform_driver simtec_audio_tlv320aic23_platdrv = { + .driver = { + .owner = THIS_MODULE, + .name = "s3c24xx-simtec-tlv320aic23", + .pm = simtec_audio_pm, + }, + .probe = simtec_audio_tlv320aic23_probe, + .remove = __devexit_p(simtec_audio_remove), +}; + +MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); + +static int __init simtec_tlv320aic23_modinit(void) +{ + return platform_driver_register(&simtec_audio_tlv320aic23_platdrv); +} + +static void __exit simtec_tlv320aic23_modexit(void) +{ + platform_driver_unregister(&simtec_audio_tlv320aic23_platdrv); +} + +module_init(simtec_tlv320aic23_modinit); +module_exit(simtec_tlv320aic23_modexit); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c new file mode 100644 index 0000000..052d596 --- /dev/null +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -0,0 +1,373 @@ +/* + * Modifications by Christian Pellegrin <chripell(a)evolware.org> + * + * s3c24xx_uda134x.c -- S3C24XX_UDA134X ALSA SoC Audio board driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/mutex.h> +#include <linux/gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/s3c24xx_uda134x.h> +#include <sound/uda134x.h> + +#include <plat/regs-iis.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "../codecs/uda134x.h" + + +/* #define ENFORCE_RATES 1 */ +/* + Unfortunately the S3C24XX in master mode has a limited capacity of + generating the clock for the codec. If you define this only rates + that are really available will be enforced. But be careful, most + user level application just want the usual sampling frequencies (8, + 11.025, 22.050, 44.1 kHz) and anyway resampling is a costly + operation for embedded systems. So if you aren't very lucky or your + hardware engineer wasn't very forward-looking it's better to leave + this undefined. If you do so an approximate value for the requested + sampling rate in the range -/+ 5% will be chosen. If this in not + possible an error will be returned. +*/ + +static struct clk *xtal; +static struct clk *pclk; +/* this is need because we don't have a place where to keep the + * pointers to the clocks in each substream. We get the clocks only + * when we are actually using them so we don't block stuff like + * frequency change or oscillator power-off */ +static int clk_users; +static DEFINE_MUTEX(clk_lock); + +static unsigned int rates[33 * 2]; +#ifdef ENFORCE_RATES +static struct snd_pcm_hw_constraint_list hw_constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; +#endif + +static struct platform_device *s3c24xx_uda134x_snd_device; + +static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) +{ + int ret = 0; +#ifdef ENFORCE_RATES + struct snd_pcm_runtime *runtime = substream->runtime; +#endif + + mutex_lock(&clk_lock); + pr_debug("%s %d\n", __func__, clk_users); + if (clk_users == 0) { + xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal"); + if (!xtal) { + printk(KERN_ERR "%s cannot get xtal\n", __func__); + ret = -EBUSY; + } else { + pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, + "pclk"); + if (!pclk) { + printk(KERN_ERR "%s cannot get pclk\n", + __func__); + clk_put(xtal); + ret = -EBUSY; + } + } + if (!ret) { + int i, j; + + for (i = 0; i < 2; i++) { + int fs = i ? 256 : 384; + + rates[i*33] = clk_get_rate(xtal) / fs; + for (j = 1; j < 33; j++) + rates[i*33 + j] = clk_get_rate(pclk) / + (j * fs); + } + } + } + clk_users += 1; + mutex_unlock(&clk_lock); + if (!ret) { +#ifdef ENFORCE_RATES + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &hw_constraints_rates); + if (ret < 0) + printk(KERN_ERR "%s cannot set constraints\n", + __func__); +#endif + } + return ret; +} + +static void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream) +{ + mutex_lock(&clk_lock); + pr_debug("%s %d\n", __func__, clk_users); + clk_users -= 1; + if (clk_users == 0) { + clk_put(xtal); + xtal = NULL; + clk_put(pclk); + pclk = NULL; + } + mutex_unlock(&clk_lock); +} + +static int s3c24xx_uda134x_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; + unsigned int clk = 0; + int ret = 0; + int clk_source, fs_mode; + unsigned long rate = params_rate(params); + long err, cerr; + unsigned int div; + int i, bi; + + err = 999999; + bi = 0; + for (i = 0; i < 2*33; i++) { + cerr = rates[i] - rate; + if (cerr < 0) + cerr = -cerr; + if (cerr < err) { + err = cerr; + bi = i; + } + } + if (bi / 33 == 1) + fs_mode = S3C2410_IISMOD_256FS; + else + fs_mode = S3C2410_IISMOD_384FS; + if (bi % 33 == 0) { + clk_source = S3C24XX_CLKSRC_MPLL; + div = 1; + } else { + clk_source = S3C24XX_CLKSRC_PCLK; + div = bi % 33; + } + pr_debug("%s desired rate %lu, %d\n", __func__, rate, bi); + + clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate; + pr_debug("%s will use: %s %s %d sysclk %d err %ld\n", __func__, + fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS", + clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK", + div, clk, err); + + if ((err * 100 / rate) > 5) { + printk(KERN_ERR "S3C24XX_UDA134X: effective frequency " + "too different from desired (%ld%%)\n", + err * 100 / rate); + return -EINVAL; + } + + 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; + + 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; + + ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source , clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, fs_mode); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div, div)); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops s3c24xx_uda134x_ops = { + .startup = s3c24xx_uda134x_startup, + .shutdown = s3c24xx_uda134x_shutdown, + .hw_params = s3c24xx_uda134x_hw_params, +}; + +static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { + .name = "UDA134X", + .stream_name = "UDA134X", + .codec_dai = &uda134x_dai, + .cpu_dai = &s3c24xx_i2s_dai, + .ops = &s3c24xx_uda134x_ops, +}; + +static struct snd_soc_card snd_soc_s3c24xx_uda134x = { + .name = "S3C24XX_UDA134X", + .platform = &s3c24xx_soc_platform, + .dai_link = &s3c24xx_uda134x_dai_link, + .num_links = 1, +}; + +static struct s3c24xx_uda134x_platform_data *s3c24xx_uda134x_l3_pins; + +static void setdat(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_data, v > 0); +} + +static void setclk(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_clk, v > 0); +} + +static void setmode(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_mode, v > 0); +} + +static struct uda134x_platform_data s3c24xx_uda134x = { + .l3 = { + .setdat = setdat, + .setclk = setclk, + .setmode = setmode, + .data_hold = 1, + .data_setup = 1, + .clock_high = 1, + .mode_hold = 1, + .mode = 1, + .mode_setup = 1, + }, +}; + +static struct snd_soc_device s3c24xx_uda134x_snd_devdata = { + .card = &snd_soc_s3c24xx_uda134x, + .codec_dev = &soc_codec_dev_uda134x, + .codec_data = &s3c24xx_uda134x, +}; + +static int s3c24xx_uda134x_setup_pin(int pin, char *fun) +{ + if (gpio_request(pin, "s3c24xx_uda134x") < 0) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "l3 %s pin already in use", fun); + return -EBUSY; + } + gpio_direction_output(pin, 0); + return 0; +} + +static int s3c24xx_uda134x_probe(struct platform_device *pdev) +{ + int ret; + + printk(KERN_INFO "S3C24XX_UDA134X SoC Audio driver\n"); + + s3c24xx_uda134x_l3_pins = pdev->dev.platform_data; + if (s3c24xx_uda134x_l3_pins == NULL) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "unable to find platform data\n"); + return -ENODEV; + } + s3c24xx_uda134x.power = s3c24xx_uda134x_l3_pins->power; + s3c24xx_uda134x.model = s3c24xx_uda134x_l3_pins->model; + + if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_data, + "data") < 0) + return -EBUSY; + if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_clk, + "clk") < 0) { + gpio_free(s3c24xx_uda134x_l3_pins->l3_data); + return -EBUSY; + } + if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_mode, + "mode") < 0) { + gpio_free(s3c24xx_uda134x_l3_pins->l3_data); + gpio_free(s3c24xx_uda134x_l3_pins->l3_clk); + return -EBUSY; + } + + s3c24xx_uda134x_snd_device = platform_device_alloc("soc-audio", -1); + if (!s3c24xx_uda134x_snd_device) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "Unable to register\n"); + return -ENOMEM; + } + + platform_set_drvdata(s3c24xx_uda134x_snd_device, + &s3c24xx_uda134x_snd_devdata); + s3c24xx_uda134x_snd_devdata.dev = &s3c24xx_uda134x_snd_device->dev; + ret = platform_device_add(s3c24xx_uda134x_snd_device); + if (ret) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n"); + platform_device_put(s3c24xx_uda134x_snd_device); + } + + return ret; +} + +static int s3c24xx_uda134x_remove(struct platform_device *pdev) +{ + platform_device_unregister(s3c24xx_uda134x_snd_device); + gpio_free(s3c24xx_uda134x_l3_pins->l3_data); + gpio_free(s3c24xx_uda134x_l3_pins->l3_clk); + gpio_free(s3c24xx_uda134x_l3_pins->l3_mode); + return 0; +} + +static struct platform_driver s3c24xx_uda134x_driver = { + .probe = s3c24xx_uda134x_probe, + .remove = s3c24xx_uda134x_remove, + .driver = { + .name = "s3c24xx_uda134x", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c24xx_uda134x_init(void) +{ + return platform_driver_register(&s3c24xx_uda134x_driver); +} + +static void __exit s3c24xx_uda134x_exit(void) +{ + platform_driver_unregister(&s3c24xx_uda134x_driver); +} + + +module_init(s3c24xx_uda134x_init); +module_exit(s3c24xx_uda134x_exit); + +MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>"); +MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c64xx-i2s-v4.c b/sound/soc/samsung/s3c64xx-i2s-v4.c new file mode 100644 index 0000000..06db130 --- /dev/null +++ b/sound/soc/samsung/s3c64xx-i2s-v4.c @@ -0,0 +1,209 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s-v4.c + * + * ALSA SoC Audio Layer - S3C64XX I2Sv4 driver + * Copyright (c) 2010 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.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. + */ + +#include <linux/clk.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include <mach/gpio-bank-c.h> +#include <mach/gpio-bank-h.h> +#include <plat/gpio-cfg.h> + +#include <mach/map.h> +#include <mach/dma.h> + +#include "s3c-dma.h" +#include "regs-i2s-v2.h" +#include "s3c64xx-i2s.h" + +static struct s3c2410_dma_client s3c64xx_dma_client_out = { + .name = "I2Sv4 PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c64xx_dma_client_in = { + .name = "I2Sv4 PCM Stereo in" +}; + +static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_out; +static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_in; +static struct s3c_i2sv2_info s3c64xx_i2sv4; + +struct snd_soc_dai s3c64xx_i2s_v4_dai; +EXPORT_SYMBOL_GPL(s3c64xx_i2s_v4_dai); + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static int s3c64xx_i2sv4_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + /* configure GPIO for i2s port */ + s3c_gpio_cfgpin(S3C64XX_GPC(4), S3C64XX_GPC4_I2S_V40_DO0); + s3c_gpio_cfgpin(S3C64XX_GPC(5), S3C64XX_GPC5_I2S_V40_DO1); + s3c_gpio_cfgpin(S3C64XX_GPC(7), S3C64XX_GPC7_I2S_V40_DO2); + s3c_gpio_cfgpin(S3C64XX_GPH(6), S3C64XX_GPH6_I2S_V40_BCLK); + s3c_gpio_cfgpin(S3C64XX_GPH(7), S3C64XX_GPH7_I2S_V40_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPH(8), S3C64XX_GPH8_I2S_V40_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPH(9), S3C64XX_GPH9_I2S_V40_DI); + + return 0; +} + +static int s3c_i2sv4_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + struct s3c_dma_params *dma_data; + u32 iismod; + + dev_dbg(cpu_dai->dev, "Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = i2s->dma_playback; + else + dma_data = i2s->dma_capture; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + + iismod = readl(i2s->regs + S3C2412_IISMOD); + dev_dbg(cpu_dai->dev, "%s: r: IISMOD: %x\n", __func__, iismod); + + iismod &= ~S3C64XX_IISMOD_BLC_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod |= S3C64XX_IISMOD_BLC_8BIT; + break; + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S24_LE: + iismod |= S3C64XX_IISMOD_BLC_24BIT; + break; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + dev_dbg(cpu_dai->dev, "%s: w: IISMOD: %x\n", __func__, iismod); + + return 0; +} + +static struct snd_soc_dai_ops s3c64xx_i2sv4_dai_ops = { + .hw_params = s3c_i2sv4_hw_params, +}; + +static __devinit int s3c64xx_i2sv4_dev_probe(struct platform_device *pdev) +{ + struct s3c_i2sv2_info *i2s; + struct snd_soc_dai *dai; + int ret; + + i2s = &s3c64xx_i2sv4; + dai = &s3c64xx_i2s_v4_dai; + + if (dai->dev) { + dev_dbg(dai->dev, "%s: \ + I2Sv4 instance already registered!\n", __func__); + return -EBUSY; + } + + dai->dev = &pdev->dev; + dai->name = "s3c64xx-i2s-v4"; + dai->id = 0; + dai->symmetric_rates = 1; + dai->playback.channels_min = 2; + dai->playback.channels_max = 2; + dai->playback.rates = S3C64XX_I2S_RATES; + dai->playback.formats = S3C64XX_I2S_FMTS; + dai->capture.channels_min = 2; + dai->capture.channels_max = 2; + dai->capture.rates = S3C64XX_I2S_RATES; + dai->capture.formats = S3C64XX_I2S_FMTS; + dai->probe = s3c64xx_i2sv4_probe; + dai->ops = &s3c64xx_i2sv4_dai_ops; + + i2s->feature |= S3C_FEATURE_CDCLKCON; + + i2s->dma_capture = &s3c64xx_i2sv4_pcm_stereo_in; + i2s->dma_playback = &s3c64xx_i2sv4_pcm_stereo_out; + + i2s->dma_capture->channel = DMACH_HSI_I2SV40_RX; + i2s->dma_capture->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISRXD; + i2s->dma_playback->channel = DMACH_HSI_I2SV40_TX; + i2s->dma_playback->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISTXD; + + i2s->dma_capture->client = &s3c64xx_dma_client_in; + i2s->dma_capture->dma_size = 4; + i2s->dma_playback->client = &s3c64xx_dma_client_out; + i2s->dma_playback->dma_size = 4; + + i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); + if (IS_ERR(i2s->iis_cclk)) { + dev_err(&pdev->dev, "failed to get audio-bus\n"); + ret = PTR_ERR(i2s->iis_cclk); + goto err; + } + + clk_enable(i2s->iis_cclk); + + ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); + if (ret) + goto err_clk; + + ret = s3c_i2sv2_register_dai(dai); + if (ret != 0) + goto err_i2sv2; + + return 0; + +err_i2sv2: + /* Not implemented for I2Sv2 core yet */ +err_clk: + clk_put(i2s->iis_cclk); +err: + return ret; +} + +static __devexit int s3c64xx_i2sv4_dev_remove(struct platform_device *pdev) +{ + dev_err(&pdev->dev, "Device removal not yet supported\n"); + return 0; +} + +static struct platform_driver s3c64xx_i2sv4_driver = { + .probe = s3c64xx_i2sv4_dev_probe, + .remove = s3c64xx_i2sv4_dev_remove, + .driver = { + .name = "s3c64xx-iis-v4", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c64xx_i2sv4_init(void) +{ + return platform_driver_register(&s3c64xx_i2sv4_driver); +} +module_init(s3c64xx_i2sv4_init); + +static void __exit s3c64xx_i2sv4_exit(void) +{ + platform_driver_unregister(&s3c64xx_i2sv4_driver); +} +module_exit(s3c64xx_i2sv4_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_DESCRIPTION("S3C64XX I2Sv4 SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c64xx-i2s.c b/sound/soc/samsung/s3c64xx-i2s.c new file mode 100644 index 0000000..1d85cb8 --- /dev/null +++ b/sound/soc/samsung/s3c64xx-i2s.c @@ -0,0 +1,190 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s.c + * + * ALSA SoC Audio Layer - S3C64XX I2S driver + * + * Copyright 2008 Openmoko, Inc. + * Copyright 2008 Simtec Electronics + * Ben Dooks <ben(a)simtec.co.uk> + * http://armlinux.simtec.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 + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/soc.h> + +#include <mach/gpio-bank-d.h> +#include <mach/gpio-bank-e.h> +#include <plat/gpio-cfg.h> + +#include <mach/map.h> +#include <mach/dma.h> + +#include "s3c-dma.h" +#include "regs-i2s-v2.h" +#include "s3c64xx-i2s.h" + +/* The value should be set to maximum of the total number + * of I2Sv3 controllers that any supported SoC has. + */ +#define MAX_I2SV3 2 + +static struct s3c2410_dma_client s3c64xx_dma_client_out = { + .name = "I2S PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c64xx_dma_client_in = { + .name = "I2S PCM Stereo in" +}; + +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_out[MAX_I2SV3]; +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_in[MAX_I2SV3]; +static struct s3c_i2sv2_info s3c64xx_i2s[MAX_I2SV3]; + +struct snd_soc_dai s3c64xx_i2s_dai[MAX_I2SV3]; +EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static int s3c64xx_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + /* configure GPIO for i2s port */ + switch (dai->id) { + case 0: + s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK); + s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPD(2), S3C64XX_GPD2_I2S0_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPD(3), S3C64XX_GPD3_I2S0_DI); + s3c_gpio_cfgpin(S3C64XX_GPD(4), S3C64XX_GPD4_I2S0_D0); + break; + case 1: + s3c_gpio_cfgpin(S3C64XX_GPE(0), S3C64XX_GPE0_I2S1_CLK); + s3c_gpio_cfgpin(S3C64XX_GPE(1), S3C64XX_GPE1_I2S1_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPE(2), S3C64XX_GPE2_I2S1_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPE(3), S3C64XX_GPE3_I2S1_DI); + s3c_gpio_cfgpin(S3C64XX_GPE(4), S3C64XX_GPE4_I2S1_D0); + } + + return 0; +} + + +static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops; + +static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) +{ + struct s3c_i2sv2_info *i2s; + struct snd_soc_dai *dai; + int ret; + + if (pdev->id >= MAX_I2SV3) { + dev_err(&pdev->dev, "id %d out of range\n", pdev->id); + return -EINVAL; + } + + i2s = &s3c64xx_i2s[pdev->id]; + dai = &s3c64xx_i2s_dai[pdev->id]; + dai->dev = &pdev->dev; + dai->name = "s3c64xx-i2s"; + dai->id = pdev->id; + dai->symmetric_rates = 1; + dai->playback.channels_min = 2; + dai->playback.channels_max = 2; + dai->playback.rates = S3C64XX_I2S_RATES; + dai->playback.formats = S3C64XX_I2S_FMTS; + dai->capture.channels_min = 2; + dai->capture.channels_max = 2; + dai->capture.rates = S3C64XX_I2S_RATES; + dai->capture.formats = S3C64XX_I2S_FMTS; + dai->probe = s3c64xx_i2s_probe; + dai->ops = &s3c64xx_i2s_dai_ops; + + i2s->feature |= S3C_FEATURE_CDCLKCON; + + i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id]; + i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id]; + + if (pdev->id == 0) { + i2s->dma_capture->channel = DMACH_I2S0_IN; + i2s->dma_capture->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISRXD; + i2s->dma_playback->channel = DMACH_I2S0_OUT; + i2s->dma_playback->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISTXD; + } else { + i2s->dma_capture->channel = DMACH_I2S1_IN; + i2s->dma_capture->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISRXD; + i2s->dma_playback->channel = DMACH_I2S1_OUT; + i2s->dma_playback->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISTXD; + } + + i2s->dma_capture->client = &s3c64xx_dma_client_in; + i2s->dma_capture->dma_size = 4; + i2s->dma_playback->client = &s3c64xx_dma_client_out; + i2s->dma_playback->dma_size = 4; + + i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); + if (IS_ERR(i2s->iis_cclk)) { + dev_err(&pdev->dev, "failed to get audio-bus\n"); + ret = PTR_ERR(i2s->iis_cclk); + goto err; + } + + clk_enable(i2s->iis_cclk); + + ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); + if (ret) + goto err_clk; + + ret = s3c_i2sv2_register_dai(dai); + if (ret != 0) + goto err_i2sv2; + + return 0; + +err_i2sv2: + /* Not implemented for I2Sv2 core yet */ +err_clk: + clk_put(i2s->iis_cclk); +err: + return ret; +} + +static __devexit int s3c64xx_iis_dev_remove(struct platform_device *pdev) +{ + dev_err(&pdev->dev, "Device removal not yet supported\n"); + return 0; +} + +static struct platform_driver s3c64xx_iis_driver = { + .probe = s3c64xx_iis_dev_probe, + .remove = s3c64xx_iis_dev_remove, + .driver = { + .name = "s3c64xx-iis", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c64xx_i2s_init(void) +{ + return platform_driver_register(&s3c64xx_iis_driver); +} +module_init(s3c64xx_i2s_init); + +static void __exit s3c64xx_i2s_exit(void) +{ + platform_driver_unregister(&s3c64xx_iis_driver); +} +module_exit(s3c64xx_i2s_exit); + +/* Module information */ +MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("S3C64XX I2S SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c64xx-i2s.h b/sound/soc/samsung/s3c64xx-i2s.h new file mode 100644 index 0000000..7a40f43 --- /dev/null +++ b/sound/soc/samsung/s3c64xx-i2s.h @@ -0,0 +1,42 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s.h + * + * ALSA SoC Audio Layer - S3C64XX I2S driver + * + * Copyright 2008 Openmoko, Inc. + * Copyright 2008 Simtec Electronics + * Ben Dooks <ben(a)simtec.co.uk> + * http://armlinux.simtec.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 + * published by the Free Software Foundation. + */ + +#ifndef __SND_SOC_S3C24XX_S3C64XX_I2S_H +#define __SND_SOC_S3C24XX_S3C64XX_I2S_H __FILE__ + +struct clk; + +#include "s3c-i2s-v2.h" + +#define S3C64XX_DIV_BCLK S3C_I2SV2_DIV_BCLK +#define S3C64XX_DIV_RCLK S3C_I2SV2_DIV_RCLK +#define S3C64XX_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER + +#define S3C64XX_CLKSRC_PCLK S3C_I2SV2_CLKSRC_PCLK +#define S3C64XX_CLKSRC_MUX S3C_I2SV2_CLKSRC_AUDIOBUS +#define S3C64XX_CLKSRC_CDCLK S3C_I2SV2_CLKSRC_CDCLK + +#define S3C64XX_I2S_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +#define S3C64XX_I2S_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +extern struct snd_soc_dai s3c64xx_i2s_dai[]; +extern struct snd_soc_dai s3c64xx_i2s_v4_dai; + +#endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */ diff --git a/sound/soc/samsung/smdk2443_wm9710.c b/sound/soc/samsung/smdk2443_wm9710.c new file mode 100644 index 0000000..3622588 --- /dev/null +++ b/sound/soc/samsung/smdk2443_wm9710.c @@ -0,0 +1,81 @@ +/* + * smdk2443_wm9710.c -- SoC audio for smdk2443 + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/ac97.h" +#include "s3c-dma.h" +#include "s3c-ac97.h" + +static struct snd_soc_card smdk2443; + +static struct snd_soc_dai_link smdk2443_dai[] = { +{ + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], + .codec_dai = &ac97_dai, +}, +}; + +static struct snd_soc_card smdk2443 = { + .name = "SMDK2443", + .platform = &s3c24xx_soc_platform, + .dai_link = smdk2443_dai, + .num_links = ARRAY_SIZE(smdk2443_dai), +}; + +static struct snd_soc_device smdk2443_snd_ac97_devdata = { + .card = &smdk2443, + .codec_dev = &soc_codec_dev_ac97, +}; + +static struct platform_device *smdk2443_snd_ac97_device; + +static int __init smdk2443_init(void) +{ + int ret; + + smdk2443_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!smdk2443_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(smdk2443_snd_ac97_device, + &smdk2443_snd_ac97_devdata); + smdk2443_snd_ac97_devdata.dev = &smdk2443_snd_ac97_device->dev; + ret = platform_device_add(smdk2443_snd_ac97_device); + + if (ret) + platform_device_put(smdk2443_snd_ac97_device); + + return ret; +} + +static void __exit smdk2443_exit(void) +{ + platform_device_unregister(smdk2443_snd_ac97_device); +} + +module_init(smdk2443_init); +module_exit(smdk2443_exit); + +/* Module information */ +MODULE_AUTHOR("Graeme Gregory, graeme.gregory(a)wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_DESCRIPTION("ALSA SoC WM9710 SMDK2443"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/smdk64xx_wm8580.c b/sound/soc/samsung/smdk64xx_wm8580.c new file mode 100644 index 0000000..07e8e51 --- /dev/null +++ b/sound/soc/samsung/smdk64xx_wm8580.c @@ -0,0 +1,266 @@ +/* + * smdk64xx_wm8580.c + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.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/platform_device.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm8580.h" +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +/* SMDK64XX has a 12MHZ crystal attached to WM8580 */ +#define SMDK64XX_WM8580_FREQ 12000000 + +static int smdk64xx_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; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int pll_out; + int bfs, rfs, ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + bfs = 16; + break; + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + bfs = 32; + break; + default: + return -EINVAL; + } + + /* The Fvco for WM8580 PLLs must fall within [90,100]MHz. + * This criterion can't be met if we request PLL output + * as {8000x256, 64000x256, 11025x256}Hz. + * As a wayout, we rather change rfs to a minimum value that + * results in (params_rate(params) * rfs), and itself, acceptable + * to both - the CODEC and the CPU. + */ + switch (params_rate(params)) { + case 16000: + case 22050: + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + rfs = 256; + break; + case 64000: + rfs = 384; + break; + case 8000: + case 11025: + rfs = 512; + break; + default: + return -EINVAL; + } + pll_out = params_rate(params) * rfs; + + /* Set the 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 the AP 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; + + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_CDCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* We use PCLK for basic ops in SoC-Slave mode */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set WM8580 to drive MCLK from its PLLA */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, + WM8580_CLKSRC_PLLA); + if (ret < 0) + return ret; + + /* Explicitly set WM8580-DAC to source from MCLK */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL, + WM8580_CLKSRC_MCLK); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, + SMDK64XX_WM8580_FREQ, pll_out); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs); + if (ret < 0) + return ret; + + return 0; +} + +/* + * SMDK64XX WM8580 DAI operations. + */ +static struct snd_soc_ops smdk64xx_ops = { + .hw_params = smdk64xx_hw_params, +}; + +/* SMDK64xx Playback widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { + SND_SOC_DAPM_HP("Front-L/R", NULL), + SND_SOC_DAPM_HP("Center/Sub", NULL), + SND_SOC_DAPM_HP("Rear-L/R", NULL), +}; + +/* SMDK64xx Capture widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { + SND_SOC_DAPM_MIC("MicIn", NULL), + SND_SOC_DAPM_LINE("LineIn", NULL), +}; + +/* SMDK-PAIFTX connections */ +static const struct snd_soc_dapm_route audio_map_tx[] = { + /* MicIn feeds AINL */ + {"AINL", NULL, "MicIn"}, + + /* LineIn feeds AINL/R */ + {"AINL", NULL, "LineIn"}, + {"AINR", NULL, "LineIn"}, +}; + +/* SMDK-PAIFRX connections */ +static const struct snd_soc_dapm_route audio_map_rx[] = { + /* Front Left/Right are fed VOUT1L/R */ + {"Front-L/R", NULL, "VOUT1L"}, + {"Front-L/R", NULL, "VOUT1R"}, + + /* Center/Sub are fed VOUT2L/R */ + {"Center/Sub", NULL, "VOUT2L"}, + {"Center/Sub", NULL, "VOUT2R"}, + + /* Rear Left/Right are fed VOUT3L/R */ + {"Rear-L/R", NULL, "VOUT3L"}, + {"Rear-L/R", NULL, "VOUT3R"}, +}; + +static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Capture widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt, + ARRAY_SIZE(wm8580_dapm_widgets_cpt)); + + /* Set up PAIFTX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); + + /* Enabling the microphone requires the fitting of a 0R + * resistor to connect the line from the microphone jack. + */ + snd_soc_dapm_disable_pin(codec, "MicIn"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Playback widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk, + ARRAY_SIZE(wm8580_dapm_widgets_pbk)); + + /* Set up PAIFRX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link smdk64xx_dai[] = { +{ /* Primary Playback i/f */ + .name = "WM8580 PAIF RX", + .stream_name = "Playback", + .cpu_dai = &s3c64xx_i2s_v4_dai, + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], + .init = smdk64xx_wm8580_init_paifrx, + .ops = &smdk64xx_ops, +}, +{ /* Primary Capture i/f */ + .name = "WM8580 PAIF TX", + .stream_name = "Capture", + .cpu_dai = &s3c64xx_i2s_v4_dai, + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], + .init = smdk64xx_wm8580_init_paiftx, + .ops = &smdk64xx_ops, +}, +}; + +static struct snd_soc_card smdk64xx = { + .name = "smdk64xx", + .platform = &s3c24xx_soc_platform, + .dai_link = smdk64xx_dai, + .num_links = ARRAY_SIZE(smdk64xx_dai), +}; + +static struct snd_soc_device smdk64xx_snd_devdata = { + .card = &smdk64xx, + .codec_dev = &soc_codec_dev_wm8580, +}; + +static struct platform_device *smdk64xx_snd_device; + +static int __init smdk64xx_audio_init(void) +{ + int ret; + + smdk64xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!smdk64xx_snd_device) + return -ENOMEM; + + platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata); + smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev; + ret = platform_device_add(smdk64xx_snd_device); + + if (ret) + platform_device_put(smdk64xx_snd_device); + + return ret; +} +module_init(smdk64xx_audio_init); + +MODULE_AUTHOR("Jaswinder Singh, jassi.brar(a)samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/smdk_wm9713.c b/sound/soc/samsung/smdk_wm9713.c new file mode 100644 index 0000000..24fd39f --- /dev/null +++ b/sound/soc/samsung/smdk_wm9713.c @@ -0,0 +1,94 @@ +/* + * smdk_wm9713.c -- SoC audio for SMDK + * + * Copyright 2010 Samsung Electronics Co. Ltd. + * Author: Jaswinder Singh Brar <jassi.brar(a)samsung.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/device.h> +#include <sound/soc.h> + +#include "../codecs/wm9713.h" +#include "s3c-dma.h" +#include "s3c-ac97.h" + +static struct snd_soc_card smdk; + +/* + * Default CFG switch settings to use this driver: + * + * SMDK6410: Set CFG1 1-3 On, CFG2 1-4 Off + */ + +/* + Playback (HeadPhone):- + $ amixer sset 'Headphone' unmute + $ amixer sset 'Right Headphone Out Mux' 'Headphone' + $ amixer sset 'Left Headphone Out Mux' 'Headphone' + $ amixer sset 'Right HP Mixer PCM' unmute + $ amixer sset 'Left HP Mixer PCM' unmute + + Capture (LineIn):- + $ amixer sset 'Right Capture Source' 'Line' + $ amixer sset 'Left Capture Source' 'Line' +*/ + +static struct snd_soc_dai_link smdk_dai = { + .name = "AC97", + .stream_name = "AC97 PCM", + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], +}; + +static struct snd_soc_card smdk = { + .name = "SMDK", + .platform = &s3c24xx_soc_platform, + .dai_link = &smdk_dai, + .num_links = 1, +}; + +static struct snd_soc_device smdk_snd_ac97_devdata = { + .card = &smdk, + .codec_dev = &soc_codec_dev_wm9713, +}; + +static struct platform_device *smdk_snd_ac97_device; + +static int __init smdk_init(void) +{ + int ret; + + smdk_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!smdk_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(smdk_snd_ac97_device, + &smdk_snd_ac97_devdata); + smdk_snd_ac97_devdata.dev = &smdk_snd_ac97_device->dev; + + ret = platform_device_add(smdk_snd_ac97_device); + if (ret) + platform_device_put(smdk_snd_ac97_device); + + return ret; +} + +static void __exit smdk_exit(void) +{ + platform_device_unregister(smdk_snd_ac97_device); +} + +module_init(smdk_init); +module_exit(smdk_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh Brar, jassi.brar(a)samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK+WM9713"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

2 1

[alsa-devel] Writing a driver for the Blackfin BF537 with two CS42448 on SPORT1
by Adam Rosenberg 25 Jun '10

25 Jun '10

I am writing my first ALSA driver for uClinux on Blackfin BF537. I have a custom board with two Cirrus Logic CS42448 chips connected to the primary and secondary data lines of SPORT1. I was basing my driver off of the driver for the Analog Devices AD1836 but it does not look like that driver was written using the latest practices. I already have the code modified well enough to communicate with both chips and produce some noise, but it is obvious that I am not filling the DMA buffer properly. Please let me know where I can find a driver that would be a better example for me to reference. When I have completed the driver I will also need help on where to submit it so that it will be supported by ALSA for others. Thank you for your help, Adam Adam Rosenberg Software Engineer Alcorn McBride Inc. 3300 South Hiawassee Building 105 Orlando, FL 32835 (407) 296 - 5800 ext. 5490

2 1

[alsa-devel] [PATCH] eukrea-tlv320: add support for our i.MX25 board
by Eric Bénard 25 Jun '10

25 Jun '10

* tdm slot has to be configured to get sound working on i.MX25 Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- sound/soc/imx/Kconfig | 2 +- sound/soc/imx/eukrea-tlv320.c | 3 ++- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index fd292f1..fd616ae 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -31,7 +31,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 bool "Eukrea TLV320" - depends on MACH_EUKREA_MBIMX27_BASEBOARD + depends on MACH_EUKREA_MBIMX27_BASEBOARD || MACH_EUKREA_MBIMXSD_BASEBOARD select SND_MXC_SOC_SSI select SND_SOC_TLV320AIC23 help diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index 968380a..45f5e4b 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -60,6 +60,7 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, pr_err("%s: failed setting codec sysclk\n", __func__); return ret; } + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, SND_SOC_CLOCK_IN); @@ -100,7 +101,7 @@ static int __init eukrea_tlv320_init(void) { int ret; - if (!machine_is_eukrea_cpuimx27()) + if (!machine_is_eukrea_cpuimx27() && !machine_is_eukrea_cpuimx25sd()) /* return happy. We might run on a totally different machine */ return 0; -- 1.6.3.3

3 13

[alsa-devel] [PATCH 1/2] ASoC: uda134x: replace a macro with a value in platform struct.
by Vladimir Zapolskiy 25 Jun '10

25 Jun '10

This change wipes out a hardcoded macro, which enables codec bias level control. Now is_powered_on_standby value shall be used instead. Signed-off-by: Vladimir Zapolskiy <vzapolskiy(a)gmail.com> --- include/sound/uda134x.h | 12 ++++++++++++ sound/soc/codecs/uda134x.c | 21 +++++---------------- 2 files changed, 17 insertions(+), 16 deletions(-) diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h index 509efb0..e475659 100644 --- a/include/sound/uda134x.h +++ b/include/sound/uda134x.h @@ -18,6 +18,18 @@ struct uda134x_platform_data { struct l3_pins l3; void (*power) (int); int model; + /* + ALSA SOC usually puts the device in standby mode when it's not used + for sometime. If you unset is_powered_on_standby the driver will + turn off the ADC/DAC when this callback is invoked and turn it back + on when needed. Unfortunately this will result in a very light bump + (it can be audible only with good earphones). If this bothers you + set is_powered_on_standby, you will have slightly higher power + consumption. Please note that sending the L3 command for ADC is + enough to make the bump, so it doesn't make difference if you + completely take off power from the codec. + */ + int is_powered_on_standby; #define UDA134X_UDA1340 1 #define UDA134X_UDA1341 2 #define UDA134X_UDA1344 3 diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index e8066b6..5c7b054 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -27,19 +27,6 @@ #include "uda134x.h" -#define POWER_OFF_ON_STANDBY 1 -/* - ALSA SOC usually puts the device in standby mode when it's not used - for sometime. If you define POWER_OFF_ON_STANDBY the driver will - turn off the ADC/DAC when this callback is invoked and turn it back - on when needed. Unfortunately this will result in a very light bump - (it can be audible only with good earphones). If this bothers you - just comment this line, you will have slightly higher power - consumption . Please note that sending the L3 command for ADC is - enough to make the bump, so it doesn't make difference if you - completely take off power from the codec. - */ - #define UDA134X_RATES SNDRV_PCM_RATE_8000_48000 #define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE) @@ -559,9 +546,11 @@ static int uda134x_soc_probe(struct platform_device *pdev) codec->num_dai = 1; codec->read = uda134x_read_reg_cache; codec->write = uda134x_write; -#ifdef POWER_OFF_ON_STANDBY - codec->set_bias_level = uda134x_set_bias_level; -#endif + + if (!pd->is_powered_on_standby) { + codec->set_bias_level = uda134x_set_bias_level; + } + INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); -- 1.7.0.4

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Re: [alsa-devel] redirecting audio to another desktop machine
by Hans-Kristian Arntzen 25 Jun '10

25 Jun '10

On 06/21/2010 12:00 PM, alsa-devel-request(a)alsa-project.org wrote: > Date: Mon, 21 Jun 2010 14:57:00 +0530 > From: Viral Mehta<Viral.Mehta(a)lntinfotech.com> > Subject: [alsa-devel] redirecting audio to another desktop machine > To: "alsa-devel(a)alsa-project.org"<alsa-devel(a)alsa-project.org> > Message-ID: > <70376CA23424B34D86F1C7DE6B9973430254343B3E(a)VSHINMSMBX01.vshodc.lntinfotech.com> > > Content-Type: text/plain; charset="us-ascii" > > Hi list, > > I have an embedded device running linux kernel and a minimal filesystem. > > I want to access this device remotely which means I should be able to connect to the device just like we do Remote Desktop over ethernet. > I am able to do this. I am able to send key-event and framebuffer data whenever my desktop machine connects to the device. > > But, I am not able to send audio data on remote side. > I am really not sure which device file (/dev/snd/ ???) I should read and capture the data and should send across.... > Or can I read the playback device ? I think NO. > Do I need to create some kind of virtual sound device ? > > Any pointers/suggestions are appreciated. > > Thanks, > Viral > > > If you want to reroute an ALSA application over the network, you can use output plugins. Afaik, both PulseAudio and RSound provides plugins for this. You generally want to set up a virtual device in .asoundrc or similar. -- Hans-Kristian Arntzen

3 5

[alsa-devel] AndreaElectronics PureAudio USB-SA
by rmfc 25 Jun '10

25 Jun '10

Hello all, I'm trying to use any USB sound card with some DSP processing that allows a microphone array to capture audio and use in speech recognition (mostly applied in noisy environments). The best device that I found so far is AndreaElectronics PureAudio USB-SA but unfortunately it isn't properly supported in Linux/Alsa. The device is recognized as an USB Audio device (snd_usb_audio) and I can record audio in Audacity. The problem is that none of DSP features is explicitly activated in USB configuration so it works almost like any other microphone. I then tried Asus Xonar U1 but the problem is the same, no configuration is done in such devices, so they record audio without enough quality to be used by a speech recognizer (not enough energy, audio DC components when software gain is applied in Sound Preferences...). Is there any way to configure an USB Audio device interfaces? Here's the output of "lsusb" of AndreaElectronics USB-SA sound card. ------------------------------------------------------------ Bus 003 Device 002: ID 08a8:000c Andrea Electronics Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x08a8 Andrea Electronics idProduct 0x000c bcdDevice 1.00 iManufacturer 3 Andrea Electronics iProduct 1 Andrea PureAudio USB-SA iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 293 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 128 bInCollection 2 baInterfaceNr( 0) 1 baInterfaceNr( 1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0603 Line Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 6 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 9 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 8 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 5 (SELECTOR_UNIT) bUnitID 8 bNrInPins 3 baSource( 0) 10 baSource( 1) 11 baSource( 2) 12 iSelector 0 AudioControl Interface Descriptor: bLength 13 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 9 bSourceID 1 bControlSize 2 bmaControls( 0) 0x01 bmaControls( 0) 0x02 Mute Loudness bmaControls( 1) 0x02 bmaControls( 1) 0x00 Volume bmaControls( 2) 0x02 bmaControls( 2) 0x00 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 10 bSourceID 2 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 11 bSourceID 3 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 12 bSourceID 4 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x00 bmaControls( 2) 0x00 iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 44100 tSamFreq[ 3] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 1 Milliseconds wLockDelay 1 Milliseconds Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 44100 tSamFreq[ 3] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 50 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x87 EP 7 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 1 Device Status: 0x0000 (Bus Powered) ------------------------------------------------------------ Renato

2 1

[alsa-devel] [PATCH] ASoC: DaVinci: Added support for stereo I2S
by Raffaele Recalcati 25 Jun '10

25 Jun '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Added audio playback support with [frame sync master - clock master] mode and with [frame sync master - clock slave]. Clock slave can be important when the external codec need system clock and bit clock synchronized. In the clock master case there is a FIXME message in the source code, because we (Davide and myself) have guessed a frequency of 122000000 that seems the base to be divided. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 7 ++ sound/soc/davinci/davinci-i2s.c | 141 ++++++++++++++++++++++++++---- 2 files changed, 129 insertions(+), 19 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 834725f..b1faeb9 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -63,6 +63,13 @@ struct snd_platform_data { unsigned sram_size_playback; unsigned sram_size_capture; + /* + * This define works when both clock and FS are output for the cpu + * and makes clock very fast (FS is not simmetrical, but sampling + * frequency is better approximated + */ + int i2s_fast_clock; + /* McASP specific fields */ int tdm_slots; u8 op_mode; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index adadcd3..8811d25 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -68,16 +68,23 @@ #define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_RCR_RFIG (1 << 18) #define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_RCR_RFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_RCR_RPHASE (1 << 31) #define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5) #define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8) #define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_XCR_XFIG (1 << 18) #define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_XCR_XFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_XCR_XPHASE (1 << 31) + +#define CLKGDV(v) (v) /* Bits 0:7 */ #define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8) #define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16) #define DAVINCI_MCBSP_SRGR_FSGM (1 << 28) +#define DAVINCI_MCBSP_SRGR_CLKSM (1 << 29) #define DAVINCI_MCBSP_PCR_CLKRP (1 << 0) #define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) @@ -144,8 +151,17 @@ struct davinci_mcbsp_dev { * won't end up being swapped because of the underrun. */ unsigned enable_channel_combine:1; + + int i2s_fast_clock; +}; + +struct davinci_mcbsp_data { + unsigned int fmt; + int clk_div; }; +static struct davinci_mcbsp_data mcbsp_data; + static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, int reg, u32 val) { @@ -255,24 +271,27 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int pcr; unsigned int srgr; srgr = DAVINCI_MCBSP_SRGR_FSGM | - DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | - DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); + DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | + DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); + /* Attention srgr is updated by hw_params! */ + mcbsp_data.fmt = fmt; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFM: case SND_SOC_DAIFMT_CBS_CFS: /* cpu is master */ pcr = DAVINCI_MCBSP_PCR_FSXM | - DAVINCI_MCBSP_PCR_FSRM | - DAVINCI_MCBSP_PCR_CLKXM | - DAVINCI_MCBSP_PCR_CLKRM; + DAVINCI_MCBSP_PCR_FSRM | + DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM; break; case SND_SOC_DAIFMT_CBM_CFS: /* McBSP CLKR pin is the input for the Sample Rate Generator. * McBSP FSR and FSX are driven by the Sample Rate Generator. */ pcr = DAVINCI_MCBSP_PCR_SCLKME | - DAVINCI_MCBSP_PCR_FSXM | - DAVINCI_MCBSP_PCR_FSRM; + DAVINCI_MCBSP_PCR_FSXM | + DAVINCI_MCBSP_PCR_FSRM; break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is master */ @@ -372,6 +391,17 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return 0; } +static int davinci_i2s_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + int srgr; + + mcbsp_data.clk_div = div; + return 0; +} + + static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -380,11 +410,12 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; - int mcbsp_word_length; - unsigned int rcr, xcr, srgr; + int mcbsp_word_length, master; + unsigned int rcr, xcr, srgr, clk_div, freq, framesize; u32 spcr; snd_pcm_format_t fmt; unsigned element_cnt = 1; + struct clk *clk; /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -396,12 +427,60 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); - srgr = DAVINCI_MCBSP_SRGR_FSGM; - srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + master = mcbsp_data.fmt & SND_SOC_DAIFMT_MASTER_MASK; + fmt = params_format(params); + mcbsp_word_length = asp_word_length[fmt]; - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); - srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + if (master == SND_SOC_DAIFMT_CBS_CFS) { + clk = clk_get(NULL, "pll1_sysclk6"); + if (clk) + freq = clk_get_rate(clk); + freq = 122000000; /* FIXME ask to Texas */ + if (dev->i2s_fast_clock) { + clk_div = 256; + do { + framesize = (freq / (--clk_div)) / + params->rate_num * + params->rate_den; + } while (((framesize < 33) || (framesize > 4095)) && + (clk_div)); + clk_div--; + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_CLKSM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * + 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1); + } else { + /* simmetric waveforms */ + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_CLKSM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * + 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + } + clk_div &= 0xFF; + srgr |= clk_div; + } else if (master == SND_SOC_DAIFMT_CBS_CFM) { + /* Clock given on CLKS */ + srgr = DAVINCI_MCBSP_SRGR_FSGM; + clk_div = mcbsp_data.clk_div - 1; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * 16 - 1); + clk_div &= 0xFF; + srgr |= clk_div; + } else { + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + srgr = DAVINCI_MCBSP_SRGR_FSGM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + pr_debug("%s - %d FWID set: re-read srgr = %X\n", + __func__, __LINE__, snd_interval_value(i) - 1); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); + srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + } davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); rcr = DAVINCI_MCBSP_RCR_RFIG; @@ -426,22 +505,45 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, element_cnt = 1; fmt = double_fmt[fmt]; } + if (master == SND_SOC_DAIFMT_CBS_CFS || + master == SND_SOC_DAIFMT_CBS_CFM) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(0); + rcr |= DAVINCI_MCBSP_RCR_RPHASE; + xcr |= DAVINCI_MCBSP_XCR_XPHASE; + } else { + /* FIXME ask to Texas */ + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(element_cnt - 1); + } } dma_params->acnt = dma_params->data_type = data_type[fmt]; dma_params->fifo_level = 0; mcbsp_word_length = asp_word_length[fmt]; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + + if (master == SND_SOC_DAIFMT_CBS_CFS || + master == SND_SOC_DAIFMT_CBS_CFM) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0); + } else { + /* FIXME ask to Texas */ + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + } rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | - DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); + DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); xcr |= DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | - DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length); + DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr); else davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr); + + pr_debug("%s - %d srgr=%X\n", __func__, __LINE__, srgr); + pr_debug("%s - %d xcr=%X\n", __func__, __LINE__, xcr); + pr_debug("%s - %d rcr=%X\n", __func__, __LINE__, rcr); return 0; } @@ -500,7 +602,7 @@ static struct snd_soc_dai_ops davinci_i2s_dai_ops = { .trigger = davinci_i2s_trigger, .hw_params = davinci_i2s_hw_params, .set_fmt = davinci_i2s_set_dai_fmt, - + .set_clkdiv = davinci_i2s_dai_set_clkdiv, }; struct snd_soc_dai davinci_i2s_dai = { @@ -548,6 +650,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) } if (pdata) { dev->enable_channel_combine = pdata->enable_channel_combine; + dev->i2s_fast_clock = pdata->i2s_fast_clock; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].sram_size = pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = -- 1.7.0.4

3 3

[alsa-devel] Compilation error
by lars.bjorndal＠broadpark.no 25 Jun '10

25 Jun '10

I get the following output while trying to compile the alsa driver on my Fedora 12 system. It worked well with the previous kernel version. # ./alsa-compile.sh --install File environment.base is cached. > cd /tmp/alsa-compile-script File environment.compile is cached. alsa-driver tree alsa-driver-1.0.23.31.g2379a.273.g5b97d is present. Reusing it. Use './alsa-compile.sh --clean=alsa-driver' command to refetch and rebuild. Sources unpacked to alsa-driver-1.0.23.31.g2379a.273.g5b97d > cd alsa-driver-1.0.23.31.g2379a.273.g5b97d Running make: make dep make[1]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d' make[2]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include' make -C sound prepare make[3]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include/sound' make prepare2 make[4]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include/sound' make[4]: Nothing to be done for `prepare2'. make[4]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include/sound' make[3]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include/sound' make[2]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include' ... make[2]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/misc' make[2]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/misc' make[1]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d' make -C /lib/modules/2.6.32.14-127.fc12.i686.PAE/build SUBDIRS=/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d CPP="gcc -E" CC="gcc" modules make[1]: Entering directory `/usr/src/kernels/2.6.32.14-127.fc12.i686.PAE' CC [M] /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/ua101.o In file included from /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/ua101.c:2: /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c: In function 'alloc_stream_buffers': /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c:1071: error: implicit declaration of function 'usb_alloc_coherent' /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c:1072: warning: assignment makes pointer from integer without a cast /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c: In function 'free_stream_buffers': /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c:1092: error: implicit declaration of function 'usb_free_coherent' /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c: In function 'ua101_card_free': /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c:1172: warning: unused variable 'ua' make[4]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/ua101.o] Error 1 make[3]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc] Error 2 make[2]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb] Error 2 make[1]: *** [_module_/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.32.14-127.fc12.i686.PAE' make: *** [compile] Error 2 Working directory /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d. Compilation of alsa-driver failed. Report this problem to the <alsa-devel(a)alsa-project.org> mailing list. Lars

1 0

[alsa-devel] Verifying mixer dB data/Invalid dB data from USB cards, especially Aureon 5.1 MkII
by Lennart Poettering 25 Jun '10

25 Jun '10

Heya, a while back I wrote a little tool "dbmeasure" which can be used to determine dB attenuation data for alsa mixer controls. I have now extended this little dB tools suite to include a tiny tool "dbverify" that can be used to verify the correctness of existing dB data exposed by the drivers. It should be relatively easy to use that tool, suitable even for non-guru folks who want to debug the dB information from alsa. http://git.0pointer.de/?p=dbmeasure.git git://git.0pointer.de/dbmeasure.git Just run "make" in a git checkout and then try: $ ./dbverify Aureon51MkII Master 30 200 That will verify the dB data of the "Master" control of the sound card that goes by the id "Aureon51MkII". It will compare the discrete volume steps 30 and 200: first it will play a full amplitude sine wave at mixer volume step 30, and then will change to mixer volume step 200, but attenuate the sine wave in software compensating for the volume change. So once you hear the sine wave attenuated by the hw, and once by the software. If the card's dB data is correct both sines should have the same volume. Now, playing around with this, I could verify what I already mentioned to Jarsolav elsewhere earlier: it seems that particularly USB cards seem to expose invalid dB data (at least all mine do). For example, for the above mentioned Aureon 5.1 it's really *way* off -- if you posess that card just run the line above, and listen to the difference. It's really bad. OTOH the dB data of the HDA cards I own is pretty accurate as it seems, at least I was unable to decide just by listening which sine wave was attenuated by sw, and which by hw. Getting back to the invalid dB data from the USB cards: the question is whether the USB descriptor data is badly parsed, and the dB mismatch hence systematic for USB cards, or if these cards are just crappy and include invalid data? In which case I wonder what we could do about that? The Aureon's dB range already looks suspicous, since both the max and the min dB value are < 0 (-47.69 to -1.97), so maybe we could add some heuristics to filter out data that already looks suspicous? Invalid dB data from the driver is a real problem for the "flat volume" logic in PA. We basically allow each app to control the full hw volume range individually, and then set the hw volume to the max of what all apps wanted and attenuate the other streams accordingly. On the Aureon this doesn't work at all, since the attenuation of the streams is miscalculated due to the invalid dB data. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4

8 93

[alsa-devel] [RFC 5/5] ASoC multi-component support : jack
by Liam Girdwood 24 Jun '10

24 Jun '10

Cache configuration is now stored as part of the codec driver structure. Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/soc-cache.c | 34 +++++++++++++++++----------------- 1 files changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 472af38..83cd8ed 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -19,7 +19,7 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; - if (reg >= codec->reg_cache_size) + if (reg >= codec->driver->reg_cache_size) return -1; return cache[reg]; } @@ -31,12 +31,12 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, u8 data[2]; int ret; - BUG_ON(codec->volatile_register); + BUG_ON(codec->driver->volatile_register); data[0] = (reg << 4) | ((value >> 8) & 0x000f); data[1] = value & 0x00ff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -89,7 +89,7 @@ static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; - if (reg >= codec->reg_cache_size) + if (reg >= codec->driver->reg_cache_size) return -1; return cache[reg]; } @@ -101,12 +101,12 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, u8 data[2]; int ret; - BUG_ON(codec->volatile_register); + BUG_ON(codec->driver->volatile_register); data[0] = (reg << 1) | ((value >> 8) & 0x0001); data[1] = value & 0x00ff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -161,13 +161,13 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, u8 *cache = codec->reg_cache; u8 data[2]; - BUG_ON(codec->volatile_register); + BUG_ON(codec->driver->volatile_register); reg &= 0xff; data[0] = reg; data[1] = value & 0xff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -188,7 +188,7 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, { u8 *cache = codec->reg_cache; reg &= 0xff; - if (reg >= codec->reg_cache_size) + if (reg >= codec->driver->reg_cache_size) return -1; return cache[reg]; } @@ -224,7 +224,7 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, { u16 *cache = codec->reg_cache; - if (reg >= codec->reg_cache_size || + if (reg >= codec->driver->reg_cache_size || snd_soc_codec_volatile_register(codec, reg)) { if (codec->cache_only) return -EINVAL; @@ -343,7 +343,7 @@ static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, u16 *cache = codec->reg_cache; reg &= 0xff; - if (reg >= codec->reg_cache_size) + if (reg >= codec->driver->reg_cache_size) return -1; return cache[reg]; } @@ -355,14 +355,14 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, u8 data[3]; int ret; - BUG_ON(codec->volatile_register); + BUG_ON(codec->driver->volatile_register); data[0] = (reg >> 8) & 0xff; data[1] = reg & 0xff; data[2] = value; reg &= 0xff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -451,7 +451,7 @@ static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec, { u16 *cache = codec->reg_cache; - if (reg >= codec->reg_cache_size || + if (reg >= codec->driver->reg_cache_size || snd_soc_codec_volatile_register(codec, reg)) { if (codec->cache_only) return -EINVAL; @@ -474,7 +474,7 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, data[2] = (value >> 8) & 0xff; data[3] = value & 0xff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -571,8 +571,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, return -EINVAL; } - codec->write = io_types[i].write; - codec->read = io_types[i].read; + codec->driver->write = io_types[i].write; + codec->driver->read = io_types[i].read; switch (control) { case SND_SOC_CUSTOM: -- 1.7.0.4

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