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