From b8ab63952fbbc74f139da96d2b4fb01d3fa9fe6d Mon Sep 17 00:00:00 2001 From: Andy Green Date: Sun, 15 Feb 2009 12:31:13 +0000 Subject: Copy the GTA02 support and edit out the items that are not present in the GTA03 (such as the external AMP) and bind to the S3C64XX I2S audio. Signed-off-by: Ben Dooks --- sound/soc/s3c24xx/Kconfig | 9 + sound/soc/s3c24xx/Makefile | 3 +- sound/soc/s3c24xx/om_gta03_wm8753.c | 568 ++++++++++++++++++++++++++++++++++++ 3 files changed, 579 insertions(+), 1 deletion(-) create mode 100644 sound/soc/s3c24xx/om_gta03_wm8753.c (limited to 'sound') diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index b845699dedd..11dcdfa663f 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -52,6 +52,15 @@ config SND_S3C24XX_SOC_NEO1973_GTA02_WM8753 Say Y if you want to add support for SoC audio on neo1973 gta02 with the WM8753 codec +config SND_S3C24XX_SOC_OM_GTA03_WM8753 + tristate "SoC I2S Audio support for OM GTA03 - WM8753" + depends on SND_S3C24XX_SOC && MACH_OPENMOKO_GTA03 + select SND_S3C64XX_SOC_I2S + select SND_SOC_WM8753 + help + Say Y if you want support for SoC audio on Openmoko GTA03 + with the WM8753 codec. + config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_S3C24XX_SOC && MACH_SMDK2443 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index de985410444..c1ff0e4bcde 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -20,6 +20,7 @@ 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-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o +snd-soc-om-gta03-wm8753-objs := om_gta03_wm8753.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 @@ -27,4 +28,4 @@ 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_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o - +obj-$(CONFIG_SND_S3C24XX_SOC_OM_GTA03_WM8753) += snd-soc-om-gta03-wm8753.o diff --git a/sound/soc/s3c24xx/om_gta03_wm8753.c b/sound/soc/s3c24xx/om_gta03_wm8753.c new file mode 100644 index 00000000000..60879b8bc80 --- /dev/null +++ b/sound/soc/s3c24xx/om_gta03_wm8753.c @@ -0,0 +1,568 @@ +/* + * om_gta03_wm8753.c -- SoC audio for GTA03 + * + * Based on neo1973_gta02_wm8753 + * + * Copyright 2009 Openmoko Inc + * Author: Ben Dooks + * Copyright 2007 Openmoko Inc + * Author: Graeme Gregory + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * + * 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 +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include + +#include "../codecs/wm8753.h" +#include "s3c24xx-pcm.h" +#include "s3c64xx-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 +#define NEO_STEREO_TO_HANDSET_SPK 9 + +static struct snd_soc_card om_gta03; + +static int om_gta03_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 = s3c64xx_i2s_get_clockrate(cpu_dai); + + 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; + +#if 0 + /* do not think we need to set this if the cpu is not the bitclk + * master */ + /* 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; +#endif + + /* 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, S3C64XX_DIV_PRESCALER, 4); + if (ret < 0) + return ret; + + /* codec PLL input is PCLK/4 */ + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, + iis_clkrate / 4, pll_out); + if (ret < 0) + return ret; + + return 0; +} + +static int om_gta03_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); +} + +/* + * GTA03 WM8753 HiFi DAI opserations. + */ +static struct snd_soc_ops om_gta03_hifi_ops = { + .hw_params = om_gta03_hifi_hw_params, + .hw_free = om_gta03_hifi_hw_free, +}; + +static int om_gta03_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 = s3c64xx_i2s_get_clockrate(rtd->dai->cpu_dai); + + 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, + iis_clkrate / 4, 12288000); + if (ret < 0) + return ret; + + return 0; +} + +static int om_gta03_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); +} + +static struct snd_soc_ops om_gta03_voice_ops = { + .hw_params = om_gta03_voice_hw_params, + .hw_free = om_gta03_voice_hw_free, +}; + +static int om_gta03_set_stereo_out(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int val = ucontrol->value.integer.value[0]; + + snd_soc_dapm_set_endpoint(codec, "Stereo Out", val); + snd_soc_dapm_sync(codec); + + return 0; +} + +static int om_gta03_get_stereo_out(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + snd_soc_dapm_get_endpoint(codec, "Stereo Out"); + + return 0; +} + + +static int om_gta03_set_gsm_out(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int val = ucontrol->value.integer.value[0]; + + snd_soc_dapm_set_endpoint(codec, "GSM Line Out", val); + snd_soc_dapm_sync(codec); + + return 0; +} + +static int om_gta03_get_gsm_out(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + snd_soc_dapm_get_endpoint(codec, "GSM Line Out"); + + return 0; +} + +static int om_gta03_set_gsm_in(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int val = ucontrol->value.integer.value[0]; + + snd_soc_dapm_set_endpoint(codec, "GSM Line In", val); + snd_soc_dapm_sync(codec); + + return 0; +} + +static int om_gta03_get_gsm_in(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + snd_soc_dapm_get_endpoint(codec, "GSM Line In"); + + return 0; +} + +static int om_gta03_set_headset_mic(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int val = ucontrol->value.integer.value[0]; + + snd_soc_dapm_set_endpoint(codec, "Headset Mic", val); + snd_soc_dapm_sync(codec); + + return 0; +} + +static int om_gta03_get_headset_mic(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + snd_soc_dapm_get_endpoint(codec, "Headset Mic"); + + return 0; +} + +static int om_gta03_set_handset_mic(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int val = ucontrol->value.integer.value[0]; + + snd_soc_dapm_set_endpoint(codec, "Handset Mic", val); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static int om_gta03_get_handset_mic(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + snd_soc_dapm_get_endpoint(codec, "Handset Mic"); + + return 0; +} + +static int om_gta03_set_handset_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int val = ucontrol->value.integer.value[0]; + + snd_soc_dapm_set_endpoint(codec, "Handset Spk", val); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static int om_gta03_get_handset_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + snd_soc_dapm_get_endpoint(codec, "Handset Spk"); + + return 0; +} + +static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { + SND_SOC_DAPM_LINE("Stereo 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("Handset Mic", NULL), + SND_SOC_DAPM_SPK("Handset Spk", NULL), +}; + + +/* example machine audio_mapnections */ +static const struct snd_soc_dapm_route audio_map[] = { + + {"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_om_gta03_controls[] = { + SOC_SINGLE_EXT("DAPM Stereo Out Switch", 0, 0, 1, 0, + om_gta03_get_stereo_out, + om_gta03_set_stereo_out), + SOC_SINGLE_EXT("DAPM GSM Line Out Switch", 1, 0, 1, 0, + om_gta03_get_gsm_out, + om_gta03_set_gsm_out), + SOC_SINGLE_EXT("DAPM GSM Line In Switch", 2, 0, 1, 0, + om_gta03_get_gsm_in, + om_gta03_set_gsm_in), + SOC_SINGLE_EXT("DAPM Headset Mic Switch", 3, 0, 1, 0, + om_gta03_get_headset_mic, + om_gta03_set_headset_mic), + SOC_SINGLE_EXT("DAPM Handset Mic Switch", 4, 0, 1, 0, + om_gta03_get_handset_mic, + om_gta03_set_handset_mic), + SOC_SINGLE_EXT("DAPM Handset Spk Switch", 5, 0, 1, 0, + om_gta03_get_handset_spk, + om_gta03_set_handset_spk), +}; + +/* + * This is an example machine initialisation for a wm8753 connected to a + * neo1973 GTA02. + */ +static int om_gta03_wm8753_init(struct snd_soc_codec *codec) +{ + int i, err; + + /* set up NC codec pins */ + snd_soc_dapm_set_endpoint(codec, "OUT3", 0); + snd_soc_dapm_set_endpoint(codec, "OUT4", 0); + snd_soc_dapm_set_endpoint(codec, "LINE1", 0); + snd_soc_dapm_set_endpoint(codec, "LINE2", 0); + + + /* Add neo1973 gta02 specific widgets */ + snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, + ARRAY_SIZE(wm8753_dapm_widgets)); + + /* add neo1973 gta02 specific controls */ + for (i = 0; i < ARRAY_SIZE(wm8753_om_gta03_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8753_om_gta03_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + /* set up neo1973 gta02 specific audio path audio_mapnects */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* set endpoints to default off mode */ + snd_soc_dapm_set_endpoint(codec, "Stereo Out", 0); + snd_soc_dapm_set_endpoint(codec, "GSM Line Out",0); + snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); + snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); + snd_soc_dapm_set_endpoint(codec, "Handset Mic", 0); + snd_soc_dapm_set_endpoint(codec, "Handset Spk", 0); + + 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 om_gta03_dai[] = { + { /* Hifi Playback - for similatious use with voice below */ + .name = "WM8753", + .stream_name = "WM8753 HiFi", + .cpu_dai = &s3c64xx_i2s_dai, + .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], + .init = om_gta03_wm8753_init, + .ops = &om_gta03_hifi_ops, + }, + { /* Voice via BT */ + .name = "Bluetooth", + .stream_name = "Voice", + .cpu_dai = &bt_dai, + .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], + .ops = &om_gta03_voice_ops, + }, +}; + +static struct snd_soc_card om_gta03 = { + .name = "om-gta03", + .platform = &s3c24xx_soc_platform, + .dai_link = om_gta03_dai, + .num_links = ARRAY_SIZE(om_gta03_dai), +}; + +/* Audio private data */ +static struct wm8753_setup_data soc_codec_data_wm8753_gta02 = { + .i2c_bus = 0, + .i2c_address = 0x1a, +}; + +static struct snd_soc_device om_gta03_snd_devdata = { + .card = &om_gta03, + .codec_dev = &soc_codec_dev_wm8753, + .codec_data = &soc_codec_data_wm8753_gta02, +}; + +static struct platform_device *om_gta03_snd_device; + +static int __init om_gta03_init(void) +{ + int ret; + + if (!machine_is_openmoko_gta03()) { + printk(KERN_INFO "Only GTA03 supported by ASoC driver\n"); + return -ENODEV; + } + + /* register bluetooth DAI here */ + ret = snd_soc_register_dai(&bt_dai); + if (ret) + return ret; + + om_gta03_snd_device = platform_device_alloc("soc-audio", 0); + if (!om_gta03_snd_device) + return -ENOMEM; + + platform_set_drvdata(om_gta03_snd_device, &om_gta03_snd_devdata); + om_gta03_snd_devdata.dev = &om_gta03_snd_device->dev; + ret = platform_device_add(om_gta03_snd_device); + + if (ret) { + platform_device_put(om_gta03_snd_device); + return ret; + } + + return ret; +} + +static void __exit om_gta03_exit(void) +{ + platform_device_unregister(om_gta03_snd_device); +} + +module_init(om_gta03_init); +module_exit(om_gta03_exit); + +/* Module information */ +MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org; Ben Dooks "); +MODULE_DESCRIPTION("ALSA SoC WM8753 OM GTA03"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3