| /* |
| * SoC audio for HTC Magician |
| * |
| * Copyright (c) 2006 Philipp Zabel <philipp.zabel@gmail.com> |
| * |
| * based on spitz.c, |
| * Authors: Liam Girdwood <lrg@slimlogic.co.uk> |
| * Richard Purdie <richard@openedhand.com> |
| * |
| * This program is free software; you can redistribute it and/or modify it |
| * under the terms of the GNU General Public License as published by the |
| * Free Software Foundation; either version 2 of the License, or (at your |
| * option) any later version. |
| * |
| */ |
| |
| #include <linux/module.h> |
| #include <linux/timer.h> |
| #include <linux/interrupt.h> |
| #include <linux/platform_device.h> |
| #include <linux/delay.h> |
| #include <linux/gpio.h> |
| #include <linux/i2c.h> |
| |
| #include <sound/core.h> |
| #include <sound/pcm.h> |
| #include <sound/pcm_params.h> |
| #include <sound/soc.h> |
| #include <sound/soc-dapm.h> |
| #include <sound/uda1380.h> |
| |
| #include <mach/magician.h> |
| #include <asm/mach-types.h> |
| #include "../codecs/uda1380.h" |
| #include "pxa2xx-pcm.h" |
| #include "pxa2xx-i2s.h" |
| #include "pxa-ssp.h" |
| |
| #define MAGICIAN_MIC 0 |
| #define MAGICIAN_MIC_EXT 1 |
| |
| static int magician_hp_switch; |
| static int magician_spk_switch = 1; |
| static int magician_in_sel = MAGICIAN_MIC; |
| |
| static void magician_ext_control(struct snd_soc_codec *codec) |
| { |
| if (magician_spk_switch) |
| snd_soc_dapm_enable_pin(codec, "Speaker"); |
| else |
| snd_soc_dapm_disable_pin(codec, "Speaker"); |
| if (magician_hp_switch) |
| snd_soc_dapm_enable_pin(codec, "Headphone Jack"); |
| else |
| snd_soc_dapm_disable_pin(codec, "Headphone Jack"); |
| |
| switch (magician_in_sel) { |
| case MAGICIAN_MIC: |
| snd_soc_dapm_disable_pin(codec, "Headset Mic"); |
| snd_soc_dapm_enable_pin(codec, "Call Mic"); |
| break; |
| case MAGICIAN_MIC_EXT: |
| snd_soc_dapm_disable_pin(codec, "Call Mic"); |
| snd_soc_dapm_enable_pin(codec, "Headset Mic"); |
| break; |
| } |
| |
| snd_soc_dapm_sync(codec); |
| } |
| |
| static int magician_startup(struct snd_pcm_substream *substream) |
| { |
| struct snd_soc_pcm_runtime *rtd = substream->private_data; |
| struct snd_soc_codec *codec = rtd->socdev->card->codec; |
| |
| /* check the jack status at stream startup */ |
| magician_ext_control(codec); |
| |
| return 0; |
| } |
| |
| /* |
| * Magician uses SSP port for playback. |
| */ |
| static int magician_playback_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 acps, acds, width, rate; |
| unsigned int div4 = PXA_SSP_CLK_SCDB_4; |
| int ret = 0; |
| |
| rate = params_rate(params); |
| width = snd_pcm_format_physical_width(params_format(params)); |
| |
| /* |
| * rate = SSPSCLK / (2 * width(16 or 32)) |
| * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1) |
| */ |
| switch (params_rate(params)) { |
| case 8000: |
| /* off by a factor of 2: bug in the PXA27x audio clock? */ |
| acps = 32842000; |
| switch (width) { |
| case 16: |
| /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ |
| acds = PXA_SSP_CLK_AUDIO_DIV_16; |
| break; |
| default: /* 32 */ |
| /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ |
| acds = PXA_SSP_CLK_AUDIO_DIV_8; |
| } |
| break; |
| case 11025: |
| acps = 5622000; |
| switch (width) { |
| case 16: |
| /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ |
| acds = PXA_SSP_CLK_AUDIO_DIV_4; |
| break; |
| default: /* 32 */ |
| /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ |
| acds = PXA_SSP_CLK_AUDIO_DIV_2; |
| } |
| break; |
| case 22050: |
| acps = 5622000; |
| switch (width) { |
| case 16: |
| /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ |
| acds = PXA_SSP_CLK_AUDIO_DIV_2; |
| break; |
| default: /* 32 */ |
| /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ |
| acds = PXA_SSP_CLK_AUDIO_DIV_1; |
| } |
| break; |
| case 44100: |
| acps = 5622000; |
| switch (width) { |
| case 16: |
| /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ |
| acds = PXA_SSP_CLK_AUDIO_DIV_2; |
| break; |
| default: /* 32 */ |
| /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ |
| acds = PXA_SSP_CLK_AUDIO_DIV_1; |
| } |
| break; |
| case 48000: |
| acps = 12235000; |
| switch (width) { |
| case 16: |
| /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ |
| acds = PXA_SSP_CLK_AUDIO_DIV_2; |
| break; |
| default: /* 32 */ |
| /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ |
| acds = PXA_SSP_CLK_AUDIO_DIV_1; |
| } |
| break; |
| case 96000: |
| default: |
| acps = 12235000; |
| switch (width) { |
| case 16: |
| /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ |
| acds = PXA_SSP_CLK_AUDIO_DIV_1; |
| break; |
| default: /* 32 */ |
| /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ |
| acds = PXA_SSP_CLK_AUDIO_DIV_2; |
| div4 = PXA_SSP_CLK_SCDB_1; |
| break; |
| } |
| break; |
| } |
| |
| /* set codec DAI configuration */ |
| ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB | |
| SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); |
| if (ret < 0) |
| return ret; |
| |
| /* set cpu DAI configuration */ |
| ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | |
| SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS); |
| if (ret < 0) |
| return ret; |
| |
| ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); |
| if (ret < 0) |
| return ret; |
| |
| /* set audio clock as clock source */ |
| ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, |
| SND_SOC_CLOCK_OUT); |
| if (ret < 0) |
| return ret; |
| |
| /* set the SSP audio system clock ACDS divider */ |
| ret = snd_soc_dai_set_clkdiv(cpu_dai, |
| PXA_SSP_AUDIO_DIV_ACDS, acds); |
| if (ret < 0) |
| return ret; |
| |
| /* set the SSP audio system clock SCDB divider4 */ |
| ret = snd_soc_dai_set_clkdiv(cpu_dai, |
| PXA_SSP_AUDIO_DIV_SCDB, div4); |
| if (ret < 0) |
| return ret; |
| |
| /* set SSP audio pll clock */ |
| ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, acps); |
| if (ret < 0) |
| return ret; |
| |
| return 0; |
| } |
| |
| /* |
| * Magician uses I2S for capture. |
| */ |
| static int magician_capture_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 = 0; |
| |
| /* set codec DAI configuration */ |
| ret = snd_soc_dai_set_fmt(codec_dai, |
| SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | |
| SND_SOC_DAIFMT_CBS_CFS); |
| if (ret < 0) |
| return ret; |
| |
| /* set cpu DAI configuration */ |
| ret = snd_soc_dai_set_fmt(cpu_dai, |
| SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | |
| SND_SOC_DAIFMT_CBS_CFS); |
| if (ret < 0) |
| return ret; |
| |
| /* set the I2S system clock as output */ |
| ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, |
| SND_SOC_CLOCK_OUT); |
| if (ret < 0) |
| return ret; |
| |
| return 0; |
| } |
| |
| static struct snd_soc_ops magician_capture_ops = { |
| .startup = magician_startup, |
| .hw_params = magician_capture_hw_params, |
| }; |
| |
| static struct snd_soc_ops magician_playback_ops = { |
| .startup = magician_startup, |
| .hw_params = magician_playback_hw_params, |
| }; |
| |
| static int magician_get_hp(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| ucontrol->value.integer.value[0] = magician_hp_switch; |
| return 0; |
| } |
| |
| static int magician_set_hp(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); |
| |
| if (magician_hp_switch == ucontrol->value.integer.value[0]) |
| return 0; |
| |
| magician_hp_switch = ucontrol->value.integer.value[0]; |
| magician_ext_control(codec); |
| return 1; |
| } |
| |
| static int magician_get_spk(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| ucontrol->value.integer.value[0] = magician_spk_switch; |
| return 0; |
| } |
| |
| static int magician_set_spk(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); |
| |
| if (magician_spk_switch == ucontrol->value.integer.value[0]) |
| return 0; |
| |
| magician_spk_switch = ucontrol->value.integer.value[0]; |
| magician_ext_control(codec); |
| return 1; |
| } |
| |
| static int magician_get_input(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| ucontrol->value.integer.value[0] = magician_in_sel; |
| return 0; |
| } |
| |
| static int magician_set_input(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| if (magician_in_sel == ucontrol->value.integer.value[0]) |
| return 0; |
| |
| magician_in_sel = ucontrol->value.integer.value[0]; |
| |
| switch (magician_in_sel) { |
| case MAGICIAN_MIC: |
| gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 1); |
| break; |
| case MAGICIAN_MIC_EXT: |
| gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 0); |
| } |
| |
| return 1; |
| } |
| |
| static int magician_spk_power(struct snd_soc_dapm_widget *w, |
| struct snd_kcontrol *k, int event) |
| { |
| gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, SND_SOC_DAPM_EVENT_ON(event)); |
| return 0; |
| } |
| |
| static int magician_hp_power(struct snd_soc_dapm_widget *w, |
| struct snd_kcontrol *k, int event) |
| { |
| gpio_set_value(EGPIO_MAGICIAN_EP_POWER, SND_SOC_DAPM_EVENT_ON(event)); |
| return 0; |
| } |
| |
| static int magician_mic_bias(struct snd_soc_dapm_widget *w, |
| struct snd_kcontrol *k, int event) |
| { |
| gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, SND_SOC_DAPM_EVENT_ON(event)); |
| return 0; |
| } |
| |
| /* magician machine dapm widgets */ |
| static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { |
| SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power), |
| SND_SOC_DAPM_SPK("Speaker", magician_spk_power), |
| SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias), |
| SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias), |
| }; |
| |
| /* magician machine audio_map */ |
| static const struct snd_soc_dapm_route audio_map[] = { |
| |
| /* Headphone connected to VOUTL, VOUTR */ |
| {"Headphone Jack", NULL, "VOUTL"}, |
| {"Headphone Jack", NULL, "VOUTR"}, |
| |
| /* Speaker connected to VOUTL, VOUTR */ |
| {"Speaker", NULL, "VOUTL"}, |
| {"Speaker", NULL, "VOUTR"}, |
| |
| /* Mics are connected to VINM */ |
| {"VINM", NULL, "Headset Mic"}, |
| {"VINM", NULL, "Call Mic"}, |
| }; |
| |
| static const char *input_select[] = {"Call Mic", "Headset Mic"}; |
| static const struct soc_enum magician_in_sel_enum = |
| SOC_ENUM_SINGLE_EXT(2, input_select); |
| |
| static const struct snd_kcontrol_new uda1380_magician_controls[] = { |
| SOC_SINGLE_BOOL_EXT("Headphone Switch", |
| (unsigned long)&magician_hp_switch, |
| magician_get_hp, magician_set_hp), |
| SOC_SINGLE_BOOL_EXT("Speaker Switch", |
| (unsigned long)&magician_spk_switch, |
| magician_get_spk, magician_set_spk), |
| SOC_ENUM_EXT("Input Select", magician_in_sel_enum, |
| magician_get_input, magician_set_input), |
| }; |
| |
| /* |
| * Logic for a uda1380 as connected on a HTC Magician |
| */ |
| static int magician_uda1380_init(struct snd_soc_codec *codec) |
| { |
| int err; |
| |
| /* NC codec pins */ |
| snd_soc_dapm_nc_pin(codec, "VOUTLHP"); |
| snd_soc_dapm_nc_pin(codec, "VOUTRHP"); |
| |
| /* FIXME: is anything connected here? */ |
| snd_soc_dapm_nc_pin(codec, "VINL"); |
| snd_soc_dapm_nc_pin(codec, "VINR"); |
| |
| /* Add magician specific controls */ |
| err = snd_soc_add_controls(codec, uda1380_magician_controls, |
| ARRAY_SIZE(uda1380_magician_controls)); |
| if (err < 0) |
| return err; |
| |
| /* Add magician specific widgets */ |
| snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets, |
| ARRAY_SIZE(uda1380_dapm_widgets)); |
| |
| /* Set up magician specific audio path interconnects */ |
| snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); |
| |
| snd_soc_dapm_sync(codec); |
| return 0; |
| } |
| |
| /* magician digital audio interface glue - connects codec <--> CPU */ |
| static struct snd_soc_dai_link magician_dai[] = { |
| { |
| .name = "uda1380", |
| .stream_name = "UDA1380 Playback", |
| .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], |
| .codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK], |
| .init = magician_uda1380_init, |
| .ops = &magician_playback_ops, |
| }, |
| { |
| .name = "uda1380", |
| .stream_name = "UDA1380 Capture", |
| .cpu_dai = &pxa_i2s_dai, |
| .codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE], |
| .ops = &magician_capture_ops, |
| } |
| }; |
| |
| /* magician audio machine driver */ |
| static struct snd_soc_card snd_soc_card_magician = { |
| .name = "Magician", |
| .dai_link = magician_dai, |
| .num_links = ARRAY_SIZE(magician_dai), |
| .platform = &pxa2xx_soc_platform, |
| }; |
| |
| /* magician audio subsystem */ |
| static struct snd_soc_device magician_snd_devdata = { |
| .card = &snd_soc_card_magician, |
| .codec_dev = &soc_codec_dev_uda1380, |
| }; |
| |
| static struct platform_device *magician_snd_device; |
| |
| /* |
| * FIXME: move into magician board file once merged into the pxa tree |
| */ |
| static struct uda1380_platform_data uda1380_info = { |
| .gpio_power = EGPIO_MAGICIAN_CODEC_POWER, |
| .gpio_reset = EGPIO_MAGICIAN_CODEC_RESET, |
| .dac_clk = UDA1380_DAC_CLK_WSPLL, |
| }; |
| |
| static struct i2c_board_info i2c_board_info[] = { |
| { |
| I2C_BOARD_INFO("uda1380", 0x18), |
| .platform_data = &uda1380_info, |
| }, |
| }; |
| |
| static int __init magician_init(void) |
| { |
| int ret; |
| struct i2c_adapter *adapter; |
| struct i2c_client *client; |
| |
| if (!machine_is_magician()) |
| return -ENODEV; |
| |
| adapter = i2c_get_adapter(0); |
| if (!adapter) |
| return -ENODEV; |
| client = i2c_new_device(adapter, i2c_board_info); |
| i2c_put_adapter(adapter); |
| if (!client) |
| return -ENODEV; |
| |
| ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER"); |
| if (ret) |
| goto err_request_spk; |
| ret = gpio_request(EGPIO_MAGICIAN_EP_POWER, "EP_POWER"); |
| if (ret) |
| goto err_request_ep; |
| ret = gpio_request(EGPIO_MAGICIAN_MIC_POWER, "MIC_POWER"); |
| if (ret) |
| goto err_request_mic; |
| ret = gpio_request(EGPIO_MAGICIAN_IN_SEL0, "IN_SEL0"); |
| if (ret) |
| goto err_request_in_sel0; |
| ret = gpio_request(EGPIO_MAGICIAN_IN_SEL1, "IN_SEL1"); |
| if (ret) |
| goto err_request_in_sel1; |
| |
| gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0); |
| |
| magician_snd_device = platform_device_alloc("soc-audio", -1); |
| if (!magician_snd_device) { |
| ret = -ENOMEM; |
| goto err_pdev; |
| } |
| |
| platform_set_drvdata(magician_snd_device, &magician_snd_devdata); |
| magician_snd_devdata.dev = &magician_snd_device->dev; |
| ret = platform_device_add(magician_snd_device); |
| if (ret) { |
| platform_device_put(magician_snd_device); |
| goto err_pdev; |
| } |
| |
| return 0; |
| |
| err_pdev: |
| gpio_free(EGPIO_MAGICIAN_IN_SEL1); |
| err_request_in_sel1: |
| gpio_free(EGPIO_MAGICIAN_IN_SEL0); |
| err_request_in_sel0: |
| gpio_free(EGPIO_MAGICIAN_MIC_POWER); |
| err_request_mic: |
| gpio_free(EGPIO_MAGICIAN_EP_POWER); |
| err_request_ep: |
| gpio_free(EGPIO_MAGICIAN_SPK_POWER); |
| err_request_spk: |
| return ret; |
| } |
| |
| static void __exit magician_exit(void) |
| { |
| platform_device_unregister(magician_snd_device); |
| |
| gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0); |
| gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0); |
| gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0); |
| |
| gpio_free(EGPIO_MAGICIAN_IN_SEL1); |
| gpio_free(EGPIO_MAGICIAN_IN_SEL0); |
| gpio_free(EGPIO_MAGICIAN_MIC_POWER); |
| gpio_free(EGPIO_MAGICIAN_EP_POWER); |
| gpio_free(EGPIO_MAGICIAN_SPK_POWER); |
| } |
| |
| module_init(magician_init); |
| module_exit(magician_exit); |
| |
| MODULE_AUTHOR("Philipp Zabel"); |
| MODULE_DESCRIPTION("ALSA SoC Magician"); |
| MODULE_LICENSE("GPL"); |