Linux音频驱动学习之：（2）移植wm8976声卡驱动（linux-3.4.2）
1.wm8976驱动程序：
/* * wm8976.h -- WM8976 Soc Audio driver * * 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 _WM8976_H #define _WM8976_H /* WM8976 register space */ #define WM8976_RESET 0x0 #define WM8976_POWER1 0x1 #define WM8976_POWER2 0x2 #define WM8976_POWER3 0x3 #define WM8976_IFACE 0x4 #define WM8976_COMP 0x5 #define WM8976_CLOCK 0x6 #define WM8976_ADD 0x7 #define WM8976_GPIO 0x8 #define WM8976_JACK1 0x9 #define WM8976_DAC 0xa #define WM8976_DACVOLL 0xb #define WM8976_DACVOLR 0xc #define WM8976_JACK2 0xd #define WM8976_ADC 0xe #define WM8976_ADCVOL 0xf #define WM8976_EQ1 0x12 #define WM8976_EQ2 0x13 #define WM8976_EQ3 0x14 #define WM8976_EQ4 0x15 #define WM8976_EQ5 0x16 #define WM8976_DACLIM1 0x18 #define WM8976_DACLIM2 0x19 #define WM8976_NOTCH1 0x1b #define WM8976_NOTCH2 0x1c #define WM8976_NOTCH3 0x1d #define WM8976_NOTCH4 0x1e #define WM8976_ALC1 0x20 #define WM8976_ALC2 0x21 #define WM8976_ALC3 0x22 #define WM8976_NGATE 0x23 #define WM8976_PLLN 0x24 #define WM8976_PLLK1 0x25 #define WM8976_PLLK2 0x26 #define WM8976_PLLK3 0x27 #define WM8976_3D 0x29 #define WM8976_BEEP 0x2b #define WM8976_INPUT 0x2c #define WM8976_INPPGA 0x2d #define WM8976_ADCBOOST 0x2f #define WM8976_OUTPUT 0x31 #define WM8976_MIXL 0x32 #define WM8976_MIXR 0x33 #define WM8976_HPVOLL 0x34 #define WM8976_HPVOLR 0x35 #define WM8976_SPKVOLL 0x36 #define WM8976_SPKVOLR 0x37 #define WM8976_OUT3MIX 0x38 #define WM8976_MONOMIX 0x39 #define WM8976_CACHEREGNUM 58 /* * WM8976 Clock dividers */ #define WM8976_MCLKDIV 0 #define WM8976_BCLKDIV 1 #define WM8976_OPCLKDIV 2 #define WM8976_DACOSR 3 #define WM8976_ADCOSR 4 #define WM8976_MCLKSEL 5 #define WM8976_MCLK_MCLK (0 << 8) #define WM8976_MCLK_PLL (1 << 8) #define WM8976_MCLK_DIV_1 (0 << 5) #define WM8976_MCLK_DIV_1_5 (1 << 5) #define WM8976_MCLK_DIV_2 (2 << 5) #define WM8976_MCLK_DIV_3 (3 << 5) #define WM8976_MCLK_DIV_4 (4 << 5) #define WM8976_MCLK_DIV_5_5 (5 << 5) #define WM8976_MCLK_DIV_6 (6 << 5) #define WM8976_BCLK_DIV_1 (0 << 2) #define WM8976_BCLK_DIV_2 (1 << 2) #define WM8976_BCLK_DIV_4 (2 << 2) #define WM8976_BCLK_DIV_8 (3 << 2) #define WM8976_BCLK_DIV_16 (4 << 2) #define WM8976_BCLK_DIV_32 (5 << 2) #define WM8976_DACOSR_64 (0 << 3) #define WM8976_DACOSR_128 (1 << 3) #define WM8976_ADCOSR_64 (0 << 3) #define WM8976_ADCOSR_128 (1 << 3) #define WM8976_OPCLK_DIV_1 (0 << 4) #define WM8976_OPCLK_DIV_2 (1 << 4) #define WM8976_OPCLK_DIV_3 (2 << 4) #define WM8976_OPCLK_DIV_4 (3 << 4) #endif
/* * wm8976.c -- WM8976 ALSA Soc Audio driver * * Copyright 2007-9 Wolfson Microelectronics PLC. * * 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/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/spi/spi.h> #include <linux/platform_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> #include <asm/io.h> #include "wm8976.h" static volatile unsigned int *gpbdat; static volatile unsigned int *gpbcon; /* * wm8976 register cache * We can't read the WM8976 register space when we are * using 2 wire for device control, so we cache them instead. */ static const u16 wm8976_reg[WM8976_CACHEREGNUM] = { 0x0000, 0x0000, 0x0000, 0x0000, 0x0050, 0x0000, 0x0140, 0x0000, 0x0000, 0x0000, 0x0000, 0x00ff, 0x00ff, 0x0000, 0x0100, 0x00ff, 0x00ff, 0x0000, 0x012c, 0x002c, 0x002c, 0x002c, 0x002c, 0x0000, 0x0032, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0038, 0x000b, 0x0032, 0x0000, 0x0008, 0x000c, 0x0093, 0x00e9, 0x0000, 0x0000, 0x0000, 0x0000, 0x0033, 0x0010, 0x0010, 0x0100, 0x0100, 0x0002, 0x0001, 0x0001, 0x0039, 0x0039, 0x0039, 0x0039, 0x0001, 0x0001, }; struct wm8976_priv { struct snd_soc_codec codec; u16 reg_cache[WM8976_CACHEREGNUM]; }; /* * read wm8976 register cache */ static inline unsigned int wm8976_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; if (reg == WM8976_RESET) return 0; if (reg >= WM8976_CACHEREGNUM) return -1; return cache[reg]; } /* * write wm8976 register cache */ static inline void wm8976_write_reg_cache(struct snd_soc_codec *codec, u16 reg, unsigned int value) { u16 *cache = codec->reg_cache; if (reg >= WM8976_CACHEREGNUM) return; cache[reg] = value; } static void set_csb(int val) { if (val) { *gpbdat |= (1<<2); } else { *gpbdat &= ~(1<<2); } } static void set_clk(int val) { if (val) { *gpbdat |= (1<<4); } else { *gpbdat &= ~(1<<4); } } static void set_dat(int val) { if (val) { *gpbdat |= (1<<3); } else { *gpbdat &= ~(1<<3); } } /* * write to the WM8976 register space */ static int wm8976_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { int i; unsigned short val = (reg << 9) | (value & 0x1ff); /* save */ wm8976_write_reg_cache (codec, reg, value); if ((reg == WM8976_HPVOLL) || (reg == WM8976_HPVOLR)) val |= (1<<8); /* write to register */ set_csb(1); set_dat(1); set_clk(1); for (i = 0; i < 16; i++){ if (val & (1<<15)) { set_clk(0); set_dat(1); udelay(1); set_clk(1); } else { set_clk(0); set_dat(0); udelay(1); set_clk(1); } val = val << 1; } set_csb(0); udelay(1); set_csb(1); set_dat(1); set_clk(1); return 0; } #define wm8976_reset(c) wm8976_write(c, WM8976_RESET, 0) static const char *wm8976_companding[] = {"Off", "NC", "u-law", "A-law" }; static const char *wm8976_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" }; static const char *wm8976_eqmode[] = {"Capture", "Playback" }; static const char *wm8976_bw[] = {"Narrow", "Wide" }; static const char *wm8976_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz" }; static const char *wm8976_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz" }; static const char *wm8976_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz" }; static const char *wm8976_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" }; static const char *wm8976_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz" }; static const char *wm8976_alc[] = {"ALC both on", "ALC left only", "ALC right only", "Limiter" }; static const struct soc_enum wm8976_enum[] = { SOC_ENUM_SINGLE(WM8976_COMP, 1, 4, wm8976_companding), /* adc */ SOC_ENUM_SINGLE(WM8976_COMP, 3, 4, wm8976_companding), /* dac */ SOC_ENUM_SINGLE(WM8976_DAC, 4, 4, wm8976_deemp), SOC_ENUM_SINGLE(WM8976_EQ1, 8, 2, wm8976_eqmode), SOC_ENUM_SINGLE(WM8976_EQ1, 5, 4, wm8976_eq1), SOC_ENUM_SINGLE(WM8976_EQ2, 8, 2, wm8976_bw), SOC_ENUM_SINGLE(WM8976_EQ2, 5, 4, wm8976_eq2), SOC_ENUM_SINGLE(WM8976_EQ3, 8, 2, wm8976_bw), SOC_ENUM_SINGLE(WM8976_EQ3, 5, 4, wm8976_eq3), SOC_ENUM_SINGLE(WM8976_EQ4, 8, 2, wm8976_bw), SOC_ENUM_SINGLE(WM8976_EQ4, 5, 4, wm8976_eq4), SOC_ENUM_SINGLE(WM8976_EQ5, 8, 2, wm8976_bw), SOC_ENUM_SINGLE(WM8976_EQ5, 5, 4, wm8976_eq5), SOC_ENUM_SINGLE(WM8976_ALC3, 8, 2, wm8976_alc), }; static const struct snd_kcontrol_new wm8976_snd_controls[] = { SOC_SINGLE("Digital Loopback Switch", WM8976_COMP, 0, 1, 0), SOC_ENUM("ADC Companding", wm8976_enum[0]), SOC_ENUM("DAC Companding", wm8976_enum[1]), SOC_SINGLE("Jack Detection Enable", WM8976_JACK1, 6, 1, 0), SOC_DOUBLE("DAC Inversion Switch", WM8976_DAC, 0, 1, 1, 0), //SOC_DOUBLE_R("Headphone Playback Volume", WM8976_DACVOLL, WM8976_DACVOLR, 0, 127, 0), SOC_SINGLE("High Pass Filter Switch", WM8976_ADC, 8, 1, 0), //SOC_SINGLE("High Pass Filter Switch", WM8976_ADC, 8, 1, 0), SOC_SINGLE("High Pass Cut Off", WM8976_ADC, 4, 7, 0), SOC_DOUBLE("ADC Inversion Switch", WM8976_ADC, 0, 1, 1, 0), SOC_SINGLE("Capture Volume", WM8976_ADCVOL, 0, 127, 0), SOC_ENUM("Equaliser Function", wm8976_enum[3]), SOC_ENUM("EQ1 Cut Off", wm8976_enum[4]), SOC_SINGLE("EQ1 Volume", WM8976_EQ1, 0, 31, 1), SOC_ENUM("Equaliser EQ2 Bandwith", wm8976_enum[5]), SOC_ENUM("EQ2 Cut Off", wm8976_enum[6]), SOC_SINGLE("EQ2 Volume", WM8976_EQ2, 0, 31, 1), SOC_ENUM("Equaliser EQ3 Bandwith", wm8976_enum[7]), SOC_ENUM("EQ3 Cut Off", wm8976_enum[8]), SOC_SINGLE("EQ3 Volume", WM8976_EQ3, 0, 31, 1), SOC_ENUM("Equaliser EQ4 Bandwith", wm8976_enum[9]), SOC_ENUM("EQ4 Cut Off", wm8976_enum[10]), SOC_SINGLE("EQ4 Volume", WM8976_EQ4, 0, 31, 1), SOC_ENUM("Equaliser EQ5 Bandwith", wm8976_enum[11]), SOC_ENUM("EQ5 Cut Off", wm8976_enum[12]), SOC_SINGLE("EQ5 Volume", WM8976_EQ5, 0, 31, 1), SOC_SINGLE("DAC Playback Limiter Switch", WM8976_DACLIM1, 8, 1, 0), SOC_SINGLE("DAC Playback Limiter Decay", WM8976_DACLIM1, 4, 15, 0), SOC_SINGLE("DAC Playback Limiter Attack", WM8976_DACLIM1, 0, 15, 0), SOC_SINGLE("DAC Playback Limiter Threshold", WM8976_DACLIM2, 4, 7, 0), SOC_SINGLE("DAC Playback Limiter Boost", WM8976_DACLIM2, 0, 15, 0), SOC_SINGLE("ALC Enable Switch", WM8976_ALC1, 8, 1, 0), SOC_SINGLE("ALC Capture Max Gain", WM8976_ALC1, 3, 7, 0), SOC_SINGLE("ALC Capture Min Gain", WM8976_ALC1, 0, 7, 0), SOC_SINGLE("ALC Capture ZC Switch", WM8976_ALC2, 8, 1, 0), SOC_SINGLE("ALC Capture Hold", WM8976_ALC2, 4, 7, 0), SOC_SINGLE("ALC Capture Target", WM8976_ALC2, 0, 15, 0), SOC_ENUM("ALC Capture Mode", wm8976_enum[13]), SOC_SINGLE("ALC Capture Decay", WM8976_ALC3, 4, 15, 0), SOC_SINGLE("ALC Capture Attack", WM8976_ALC3, 0, 15, 0), SOC_SINGLE("ALC Capture Noise Gate Switch", WM8976_NGATE, 3, 1, 0), SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8976_NGATE, 0, 7, 0), SOC_SINGLE("Capture PGA ZC Switch", WM8976_INPPGA, 7, 1, 0), SOC_SINGLE("Capture PGA Volume", WM8976_INPPGA, 0, 63, 0), SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8976_HPVOLL, WM8976_HPVOLR, 7, 1, 0), SOC_DOUBLE_R("Headphone Playback Switch", WM8976_HPVOLL, WM8976_HPVOLR, 6, 1, 1), SOC_DOUBLE_R("Headphone Playback Volume", WM8976_HPVOLL, WM8976_HPVOLR, 0, 63, 0), SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8976_SPKVOLL, WM8976_SPKVOLR, 7, 1, 0), SOC_DOUBLE_R("Speaker Playback Switch", WM8976_SPKVOLL, WM8976_SPKVOLR, 6, 1, 1), SOC_DOUBLE_R("Speaker Playback Volume", WM8976_SPKVOLL, WM8976_SPKVOLR, 0, 63, 0), SOC_SINGLE("Capture Boost(+20dB)", WM8976_ADCBOOST, 8, 1, 0), }; /* Left Output Mixer */ static const struct snd_kcontrol_new wm8976_left_mixer_controls[] = { SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8976_OUTPUT, 6, 1, 1), SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8976_MIXL, 0, 1, 1), SOC_DAPM_SINGLE("Line Bypass Switch", WM8976_MIXL, 1, 1, 0), SOC_DAPM_SINGLE("Aux Playback Switch", WM8976_MIXL, 5, 1, 0), }; /* Right Output Mixer */ static const struct snd_kcontrol_new wm8976_right_mixer_controls[] = { SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8976_OUTPUT, 5, 1, 1), SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8976_MIXR, 0, 1, 1), SOC_DAPM_SINGLE("Line Bypass Switch", WM8976_MIXR, 1, 1, 0), SOC_DAPM_SINGLE("Aux Playback Switch", WM8976_MIXR, 5, 1, 0), }; /* Left AUX Input boost vol */ static const struct snd_kcontrol_new wm8976_laux_boost_controls = SOC_DAPM_SINGLE("Aux Volume", WM8976_ADCBOOST, 0, 3, 0); /* Left Input boost vol */ static const struct snd_kcontrol_new wm8976_lmic_boost_controls = SOC_DAPM_SINGLE("Input Volume", WM8976_ADCBOOST, 4, 3, 0); /* Left Aux In to PGA */ static const struct snd_kcontrol_new wm8976_laux_capture_boost_controls = SOC_DAPM_SINGLE("Capture Switch", WM8976_ADCBOOST, 8, 1, 0); /* Left Input P In to PGA */ static const struct snd_kcontrol_new wm8976_lmicp_capture_boost_controls = SOC_DAPM_SINGLE("Input P Capture Boost Switch", WM8976_INPUT, 0, 1, 0); /* Left Input N In to PGA */ static const struct snd_kcontrol_new wm8976_lmicn_capture_boost_controls = SOC_DAPM_SINGLE("Input N Capture Boost Switch", WM8976_INPUT, 1, 1, 0); // TODO Widgets static const struct snd_soc_dapm_widget wm8976_dapm_widgets[] = { #if 0 //SND_SOC_DAPM_MUTE("Mono Mute", WM8976_MONOMIX, 6, 0), //SND_SOC_DAPM_MUTE("Speaker Mute", WM8976_SPKMIX, 6, 0), SND_SOC_DAPM_MIXER("Speaker Mixer", WM8976_POWER3, 2, 0, &wm8976_speaker_mixer_controls[0], ARRAY_SIZE(wm8976_speaker_mixer_controls)), SND_SOC_DAPM_MIXER("Mono Mixer", WM8976_POWER3, 3, 0, &wm8976_mono_mixer_controls[0], ARRAY_SIZE(wm8976_mono_mixer_controls)), SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8976_POWER3, 0, 0), SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8976_POWER3, 0, 0), SND_SOC_DAPM_PGA("Aux Input", WM8976_POWER1, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("SpkN Out", WM8976_POWER3, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("SpkP Out", WM8976_POWER3, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("Mono Out", WM8976_POWER3, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("Mic PGA", WM8976_POWER2, 2, 0, NULL, 0), SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0, &wm8976_aux_boost_controls, 1), SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0, &wm8976_mic_boost_controls, 1), SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0, &wm8976_capture_boost_controls), SND_SOC_DAPM_MIXER("Boost Mixer", WM8976_POWER2, 4, 0, NULL, 0), SND_SOC_DAPM_MICBIAS("Mic Bias", WM8976_POWER1, 4, 0), SND_SOC_DAPM_INPUT("MICN"), SND_SOC_DAPM_INPUT("MICP"), SND_SOC_DAPM_INPUT("AUX"), SND_SOC_DAPM_OUTPUT("MONOOUT"), SND_SOC_DAPM_OUTPUT("SPKOUTP"), SND_SOC_DAPM_OUTPUT("SPKOUTN"), #endif }; static const struct snd_soc_dapm_route audio_map[] = { /* Mono output mixer */ {"Mono Mixer", "PCM Playback Switch", "DAC"}, {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, /* Speaker output mixer */ {"Speaker Mixer", "PCM Playback Switch", "DAC"}, {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, /* Outputs */ {"Mono Out", NULL, "Mono Mixer"}, {"MONOOUT", NULL, "Mono Out"}, {"SpkN Out", NULL, "Speaker Mixer"}, {"SpkP Out", NULL, "Speaker Mixer"}, {"SPKOUTN", NULL, "SpkN Out"}, {"SPKOUTP", NULL, "SpkP Out"}, /* Boost Mixer */ {"Boost Mixer", NULL, "ADC"}, {"Capture Boost Switch", "Aux Capture Boost Switch", "AUX"}, {"Aux Boost", "Aux Volume", "Boost Mixer"}, {"Capture Boost", "Capture Switch", "Boost Mixer"}, {"Mic Boost", "Mic Volume", "Boost Mixer"}, /* Inputs */ {"MICP", NULL, "Mic Boost"}, {"MICN", NULL, "Mic PGA"}, {"Mic PGA", NULL, "Capture Boost"}, {"AUX", NULL, "Aux Input"}, }; static int wm8976_add_widgets(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = &codec->dapm; snd_soc_dapm_new_controls(dapm, wm8976_dapm_widgets, ARRAY_SIZE(wm8976_dapm_widgets)); snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_new_widgets(dapm); return 0; } struct _pll_div { unsigned int pre:4; /* prescale - 1 */ unsigned int n:4; unsigned int k; }; static struct _pll_div pll_div; /* The size in bits of the pll divide multiplied by 10 * to allow rounding later */ #define FIXED_PLL_SIZE ((1 << 24) * 10) static void pll_factors(unsigned int target, unsigned int source) { unsigned long long Kpart; unsigned int K, Ndiv, Nmod; Ndiv = target / source; if (Ndiv < 6) { source >>= 1; pll_div.pre = 1; Ndiv = target / source; } else pll_div.pre = 0; if ((Ndiv < 6) || (Ndiv > 12)) printk(KERN_WARNING "WM8976 N value outwith recommended range! N = %d\n",Ndiv); pll_div.n = Ndiv; Nmod = target % source; Kpart = FIXED_PLL_SIZE * (long long)Nmod; do_div(Kpart, source); K = Kpart & 0xFFFFFFFF; /* Check if we need to round */ if ((K % 10) >= 5) K += 5; /* Move down to proper range now rounding is done */ K /= 10; pll_div.k = K; } static int wm8976_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; u16 reg; if(freq_in == 0 || freq_out == 0) { reg = wm8976_read_reg_cache(codec, WM8976_POWER1); wm8976_write(codec, WM8976_POWER1, reg & 0x1df); return 0; } pll_factors(freq_out * 8, freq_in); wm8976_write(codec, WM8976_PLLN, (pll_div.pre << 4) | pll_div.n); wm8976_write(codec, WM8976_PLLK1, pll_div.k >> 18); wm8976_write(codec, WM8976_PLLK1, (pll_div.k >> 9) && 0x1ff); wm8976_write(codec, WM8976_PLLK1, pll_div.k && 0x1ff); reg = wm8976_read_reg_cache(codec, WM8976_POWER1); wm8976_write(codec, WM8976_POWER1, reg | 0x020); return 0; } static int wm8976_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; u16 iface = wm8976_read_reg_cache(codec, WM8976_IFACE) & 0x3; u16 clk = wm8976_read_reg_cache(codec, WM8976_CLOCK) & 0xfffe; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: clk |= 0x0001; break; case SND_SOC_DAIFMT_CBS_CFS: break; default: return -EINVAL; } /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: iface |= 0x0010; break; case SND_SOC_DAIFMT_RIGHT_J: break; case SND_SOC_DAIFMT_LEFT_J: iface |= 0x0008; break; case SND_SOC_DAIFMT_DSP_A: iface |= 0x00018; break; default: return -EINVAL; } /* clock inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: break; case SND_SOC_DAIFMT_IB_IF: iface |= 0x0180; break; case SND_SOC_DAIFMT_IB_NF: iface |= 0x0100; break; case SND_SOC_DAIFMT_NB_IF: iface |= 0x0080; break; default: return -EINVAL; } wm8976_write(codec, WM8976_IFACE, iface); wm8976_write(codec, WM8976_CLOCK, clk); return 0; } static int wm8976_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; u16 iface = wm8976_read_reg_cache(codec, WM8976_IFACE) & 0xff9f; u16 adn = wm8976_read_reg_cache(codec, WM8976_ADD) & 0x1f1; /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: break; case SNDRV_PCM_FORMAT_S20_3LE: iface |= 0x0020; break; case SNDRV_PCM_FORMAT_S24_LE: iface |= 0x0040; break; } /* filter coefficient */ switch (params_rate(params)) { case SNDRV_PCM_RATE_8000: adn |= 0x5 << 1; break; case SNDRV_PCM_RATE_11025: adn |= 0x4 << 1; break; case SNDRV_PCM_RATE_16000: adn |= 0x3 << 1; break; case SNDRV_PCM_RATE_22050: adn |= 0x2 << 1; break; case SNDRV_PCM_RATE_32000: adn |= 0x1 << 1; break; } /* set iface */ wm8976_write(codec, WM8976_IFACE, iface); wm8976_write(codec, WM8976_ADD, adn); return 0; } static int wm8976_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; u16 reg; switch (div_id) { case WM8976_MCLKDIV: reg = wm8976_read_reg_cache(codec, WM8976_CLOCK) & 0x11f; wm8976_write(codec, WM8976_CLOCK, reg | div); break; case WM8976_BCLKDIV: reg = wm8976_read_reg_cache(codec, WM8976_CLOCK) & 0x1c7; wm8976_write(codec, WM8976_CLOCK, reg | div); break; case WM8976_OPCLKDIV: reg = wm8976_read_reg_cache(codec, WM8976_GPIO) & 0x1cf; wm8976_write(codec, WM8976_GPIO, reg | div); break; case WM8976_DACOSR: reg = wm8976_read_reg_cache(codec, WM8976_DAC) & 0x1f7; wm8976_write(codec, WM8976_DAC, reg | div); break; case WM8976_ADCOSR: reg = wm8976_read_reg_cache(codec, WM8976_ADC) & 0x1f7; wm8976_write(codec, WM8976_ADC, reg | div); break; case WM8976_MCLKSEL: reg = wm8976_read_reg_cache(codec, WM8976_CLOCK) & 0x0ff; wm8976_write(codec, WM8976_CLOCK, reg | div); break; default: return -EINVAL; } return 0; } static int wm8976_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8976_read_reg_cache(codec, WM8976_DAC) & 0xffbf; if(mute) wm8976_write(codec, WM8976_DAC, mute_reg | 0x40); else { wm8976_write(codec, WM8976_DAC, mute_reg); } return 0; } /* TODO: liam need to make this lower power with dapm */ static int wm8976_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { switch (level) { case SND_SOC_BIAS_ON: wm8976_write(codec, WM8976_POWER1, 0x1ff); wm8976_write(codec, WM8976_POWER2, 0x1ff & ~(1<<6)); wm8976_write(codec, WM8976_POWER3, 0x1ff); break; case SND_SOC_BIAS_STANDBY: case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_OFF: wm8976_write(codec, WM8976_POWER1, 0x0); wm8976_write(codec, WM8976_POWER2, 0x0); wm8976_write(codec, WM8976_POWER3, 0x0); break; } codec->dapm.bias_level = level; return 0; } #define WM8976_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) #define WM8976_FORMATS \ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ SNDRV_PCM_FORMAT_S24_3LE | SNDRV_PCM_FORMAT_S24_LE) static struct snd_soc_dai_ops wm8976_dai_ops = { .hw_params = wm8976_hw_params, .digital_mute = wm8976_mute, .set_fmt = wm8976_set_dai_fmt, .set_clkdiv = wm8976_set_dai_clkdiv, .set_pll = wm8976_set_dai_pll, }; struct snd_soc_dai_driver wm8976_dai = { .name = "wm8976-iis", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = WM8976_RATES, .formats = WM8976_FORMATS,}, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 1, .rates = WM8976_RATES, .formats = WM8976_FORMATS,}, .ops = &wm8976_dai_ops, }; static int snd_soc_wm8976_suspend(struct snd_soc_codec *codec) { wm8976_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static int snd_soc_wm8976_resume(struct snd_soc_codec *codec) { int i; u16 *cache = codec->reg_cache; /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8976_reg); i++) { codec->write(codec->control_data, i, cache[i]); } wm8976_set_bias_level(codec, SND_SOC_BIAS_PREPARE); wm8976_set_bias_level(codec, SND_SOC_BIAS_ON); return 0; } static int snd_soc_wm8976_probe(struct snd_soc_codec *codec) { gpbcon = ioremap(0x56000010, 4); gpbdat = ioremap(0x56000014, 4); /* GPB 4: L3CLOCK */ /* GPB 3: L3DATA */ /* GPB 2: L3MODE */ *gpbcon &= ~((3<<4) | (3<<6) | (3<<8)); *gpbcon |= ((1<<4) | (1<<6) | (1<<8)); snd_soc_add_codec_controls(codec, wm8976_snd_controls, ARRAY_SIZE(wm8976_snd_controls)); //wm8976_add_widgets(codec); return 0; } /* power down chip */ static int snd_soc_wm8976_remove(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = &codec->dapm; //snd_soc_dapm_free(dapm); iounmap(gpbcon); iounmap(gpbdat); return 0; } struct snd_soc_codec_driver soc_codec_dev_wm8976 = { .probe = snd_soc_wm8976_probe, .remove = snd_soc_wm8976_remove, .suspend = snd_soc_wm8976_suspend, .resume = snd_soc_wm8976_resume, .reg_cache_size = sizeof(wm8976_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8976_reg, .reg_cache_step = 2, .read = wm8976_read_reg_cache, .write = wm8976_write, .set_bias_level = wm8976_set_bias_level, }; /* ͨ¹ý×¢²áƽ̨É豸¡¢Æ½Ì¨Çý¶¯À´ÊµÏÖ¶Ôsnd_soc_register_codecµÄµ÷Óà * */ static void wm8976_dev_release(struct device * dev) { } static int wm8976_probe(struct platform_device *pdev) { return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8976, &wm8976_dai, 1); } static int wm8976_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; } static struct platform_device wm8976_dev = { .name = "wm8976-codec", .id = -1, .dev = { .release = wm8976_dev_release, }, }; struct platform_driver wm8976_drv = { .probe = wm8976_probe, .remove = wm8976_remove, .driver = { .name = "wm8976-codec", } }; static int wm8976_init(void) { platform_device_register(&wm8976_dev); platform_driver_register(&wm8976_drv); return 0; } static void wm8976_exit(void) { platform_device_unregister(&wm8976_dev); platform_driver_unregister(&wm8976_drv); } module_init(wm8976_init); module_exit(wm8976_exit); MODULE_LICENSE("GPL");
#include <stdio.h> #include <stdlib.h> #include <alsa/asoundlib.h> snd_pcm_t *open_sound_dev(snd_pcm_stream_t type) { int err; snd_pcm_t *handle; snd_pcm_hw_params_t *hw_params; unsigned int rate = 44100; if ((err = snd_pcm_open (&handle, "default", type, 0)) < 0) { return NULL; } if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) { fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", snd_strerror (err)); return NULL; } if ((err = snd_pcm_hw_params_any (handle, hw_params)) < 0) { fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", snd_strerror (err)); return NULL; } if ((err = snd_pcm_hw_params_set_access (handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { fprintf (stderr, "cannot set access type (%s)\n", snd_strerror (err)); return NULL; } if ((err = snd_pcm_hw_params_set_format (handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0) { fprintf (stderr, "cannot set sample format (%s)\n", snd_strerror (err)); return NULL; } if ((err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &rate, 0)) < 0) { fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err)); return NULL; } if ((err = snd_pcm_hw_params_set_channels (handle, hw_params, 2)) < 0) { fprintf (stderr, "cannot set channel count (%s)\n", snd_strerror (err)); return NULL; } if ((err = snd_pcm_hw_params (handle, hw_params)) < 0) { fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err)); return NULL; } snd_pcm_hw_params_free (hw_params); return handle; } void close_sound_dev(snd_pcm_t *handle) { snd_pcm_close (handle); } snd_pcm_t *open_playback(void) { return open_sound_dev(SND_PCM_STREAM_PLAYBACK); } snd_pcm_t *open_capture(void) { return open_sound_dev(SND_PCM_STREAM_CAPTURE); } int main (int argc, char *argv[]) { int err; char buf[512]; snd_pcm_t *playback_handle; snd_pcm_t *capture_handle; playback_handle = open_playback(); if (!playback_handle) { fprintf (stderr, "cannot open for playback\n"); return -1; } capture_handle = open_capture(); if (!capture_handle) { fprintf (stderr, "cannot open for capture\n"); return -1; } if ((err = snd_pcm_prepare (playback_handle)) < 0) { fprintf (stderr, "cannot prepare audio interface for use (%s)\n", snd_strerror (err)); return -1; } if ((err = snd_pcm_prepare (capture_handle)) < 0) { fprintf (stderr, "cannot prepare audio interface for use (%s)\n", snd_strerror (err)); return -1; } while (1) { if ((err = snd_pcm_readi (capture_handle, buf, 128)) != 128) { fprintf (stderr, "read from audio interface failed (%s)\n", snd_strerror (err)); return -1; } if ((err = snd_pcm_writei (playback_handle, buf, 128)) != 128) { fprintf (stderr, "write to audio interface failed (%s)\n", snd_strerror (err)); return -1; } } snd_pcm_close (playback_handle); snd_pcm_close (capture_handle); return 0; }
Makefile
capture_playback : capture_playback.c arm-linux-gcc -Wall -o capture_playback capture_playback.c -lasound clean: rm capture_playback
posted on 2017-03-12 19:53 sheldon_blogs 阅读(2406) 评论(0) 编辑 收藏 举报
