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5a0e3ad6af
percpu.h is included by sched.h and module.h and thus ends up being included when building most .c files. percpu.h includes slab.h which in turn includes gfp.h making everything defined by the two files universally available and complicating inclusion dependencies. percpu.h -> slab.h dependency is about to be removed. Prepare for this change by updating users of gfp and slab facilities include those headers directly instead of assuming availability. As this conversion needs to touch large number of source files, the following script is used as the basis of conversion. http://userweb.kernel.org/~tj/misc/slabh-sweep.py The script does the followings. * Scan files for gfp and slab usages and update includes such that only the necessary includes are there. ie. if only gfp is used, gfp.h, if slab is used, slab.h. * When the script inserts a new include, it looks at the include blocks and try to put the new include such that its order conforms to its surrounding. It's put in the include block which contains core kernel includes, in the same order that the rest are ordered - alphabetical, Christmas tree, rev-Xmas-tree or at the end if there doesn't seem to be any matching order. * If the script can't find a place to put a new include (mostly because the file doesn't have fitting include block), it prints out an error message indicating which .h file needs to be added to the file. The conversion was done in the following steps. 1. The initial automatic conversion of all .c files updated slightly over 4000 files, deleting around 700 includes and adding ~480 gfp.h and ~3000 slab.h inclusions. The script emitted errors for ~400 files. 2. Each error was manually checked. Some didn't need the inclusion, some needed manual addition while adding it to implementation .h or embedding .c file was more appropriate for others. This step added inclusions to around 150 files. 3. The script was run again and the output was compared to the edits from #2 to make sure no file was left behind. 4. Several build tests were done and a couple of problems were fixed. e.g. lib/decompress_*.c used malloc/free() wrappers around slab APIs requiring slab.h to be added manually. 5. The script was run on all .h files but without automatically editing them as sprinkling gfp.h and slab.h inclusions around .h files could easily lead to inclusion dependency hell. Most gfp.h inclusion directives were ignored as stuff from gfp.h was usually wildly available and often used in preprocessor macros. Each slab.h inclusion directive was examined and added manually as necessary. 6. percpu.h was updated not to include slab.h. 7. Build test were done on the following configurations and failures were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my distributed build env didn't work with gcov compiles) and a few more options had to be turned off depending on archs to make things build (like ipr on powerpc/64 which failed due to missing writeq). * x86 and x86_64 UP and SMP allmodconfig and a custom test config. * powerpc and powerpc64 SMP allmodconfig * sparc and sparc64 SMP allmodconfig * ia64 SMP allmodconfig * s390 SMP allmodconfig * alpha SMP allmodconfig * um on x86_64 SMP allmodconfig 8. percpu.h modifications were reverted so that it could be applied as a separate patch and serve as bisection point. Given the fact that I had only a couple of failures from tests on step 6, I'm fairly confident about the coverage of this conversion patch. If there is a breakage, it's likely to be something in one of the arch headers which should be easily discoverable easily on most builds of the specific arch. Signed-off-by: Tejun Heo <tj@kernel.org> Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org> Cc: Ingo Molnar <mingo@redhat.com> Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
420 lines
11 KiB
C
420 lines
11 KiB
C
/*
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* File: sound/soc/codecs/ad1836.c
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* Author: Barry Song <Barry.Song@analog.com>
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*
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* Created: Aug 04 2009
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* Description: Driver for AD1836 sound chip
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*
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* Modified:
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* Copyright 2009 Analog Devices Inc.
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*
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* Bugs: Enter bugs at http://blackfin.uclinux.org/
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*/
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#include <linux/init.h>
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#include <linux/slab.h>
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#include <linux/module.h>
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#include <linux/kernel.h>
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#include <linux/device.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/initval.h>
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#include <sound/soc.h>
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#include <sound/tlv.h>
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#include <sound/soc-dapm.h>
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#include <linux/spi/spi.h>
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#include "ad1836.h"
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/* codec private data */
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struct ad1836_priv {
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struct snd_soc_codec codec;
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u16 reg_cache[AD1836_NUM_REGS];
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};
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static struct snd_soc_codec *ad1836_codec;
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struct snd_soc_codec_device soc_codec_dev_ad1836;
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static int ad1836_register(struct ad1836_priv *ad1836);
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static void ad1836_unregister(struct ad1836_priv *ad1836);
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/*
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* AD1836 volume/mute/de-emphasis etc. controls
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*/
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static const char *ad1836_deemp[] = {"None", "44.1kHz", "32kHz", "48kHz"};
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static const struct soc_enum ad1836_deemp_enum =
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SOC_ENUM_SINGLE(AD1836_DAC_CTRL1, 8, 4, ad1836_deemp);
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static const struct snd_kcontrol_new ad1836_snd_controls[] = {
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/* DAC volume control */
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SOC_DOUBLE_R("DAC1 Volume", AD1836_DAC_L1_VOL,
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AD1836_DAC_R1_VOL, 0, 0x3FF, 0),
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SOC_DOUBLE_R("DAC2 Volume", AD1836_DAC_L2_VOL,
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AD1836_DAC_R2_VOL, 0, 0x3FF, 0),
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SOC_DOUBLE_R("DAC3 Volume", AD1836_DAC_L3_VOL,
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AD1836_DAC_R3_VOL, 0, 0x3FF, 0),
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/* ADC switch control */
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SOC_DOUBLE("ADC1 Switch", AD1836_ADC_CTRL2, AD1836_ADCL1_MUTE,
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AD1836_ADCR1_MUTE, 1, 1),
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SOC_DOUBLE("ADC2 Switch", AD1836_ADC_CTRL2, AD1836_ADCL2_MUTE,
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AD1836_ADCR2_MUTE, 1, 1),
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/* DAC switch control */
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SOC_DOUBLE("DAC1 Switch", AD1836_DAC_CTRL2, AD1836_DACL1_MUTE,
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AD1836_DACR1_MUTE, 1, 1),
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SOC_DOUBLE("DAC2 Switch", AD1836_DAC_CTRL2, AD1836_DACL2_MUTE,
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AD1836_DACR2_MUTE, 1, 1),
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SOC_DOUBLE("DAC3 Switch", AD1836_DAC_CTRL2, AD1836_DACL3_MUTE,
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AD1836_DACR3_MUTE, 1, 1),
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/* ADC high-pass filter */
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SOC_SINGLE("ADC High Pass Filter Switch", AD1836_ADC_CTRL1,
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AD1836_ADC_HIGHPASS_FILTER, 1, 0),
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/* DAC de-emphasis */
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SOC_ENUM("Playback Deemphasis", ad1836_deemp_enum),
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};
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static const struct snd_soc_dapm_widget ad1836_dapm_widgets[] = {
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SND_SOC_DAPM_DAC("DAC", "Playback", AD1836_DAC_CTRL1,
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AD1836_DAC_POWERDOWN, 1),
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SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
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SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1836_ADC_CTRL1,
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AD1836_ADC_POWERDOWN, 1, NULL, 0),
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SND_SOC_DAPM_OUTPUT("DAC1OUT"),
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SND_SOC_DAPM_OUTPUT("DAC2OUT"),
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SND_SOC_DAPM_OUTPUT("DAC3OUT"),
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SND_SOC_DAPM_INPUT("ADC1IN"),
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SND_SOC_DAPM_INPUT("ADC2IN"),
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};
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static const struct snd_soc_dapm_route audio_paths[] = {
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{ "DAC", NULL, "ADC_PWR" },
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{ "ADC", NULL, "ADC_PWR" },
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{ "DAC1OUT", "DAC1 Switch", "DAC" },
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{ "DAC2OUT", "DAC2 Switch", "DAC" },
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{ "DAC3OUT", "DAC3 Switch", "DAC" },
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{ "ADC", "ADC1 Switch", "ADC1IN" },
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{ "ADC", "ADC2 Switch", "ADC2IN" },
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};
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/*
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* DAI ops entries
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*/
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static int ad1836_set_dai_fmt(struct snd_soc_dai *codec_dai,
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unsigned int fmt)
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{
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switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
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/* at present, we support adc aux mode to interface with
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* blackfin sport tdm mode
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*/
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case SND_SOC_DAIFMT_DSP_A:
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break;
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default:
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return -EINVAL;
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}
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switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
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case SND_SOC_DAIFMT_IB_IF:
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break;
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default:
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return -EINVAL;
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}
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switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
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/* ALCLK,ABCLK are both output, AD1836 can only be master */
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case SND_SOC_DAIFMT_CBM_CFM:
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break;
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default:
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return -EINVAL;
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}
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return 0;
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}
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static int ad1836_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params,
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struct snd_soc_dai *dai)
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{
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int word_len = 0;
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_device *socdev = rtd->socdev;
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struct snd_soc_codec *codec = socdev->card->codec;
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/* bit size */
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switch (params_format(params)) {
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case SNDRV_PCM_FORMAT_S16_LE:
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word_len = 3;
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break;
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case SNDRV_PCM_FORMAT_S20_3LE:
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word_len = 1;
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break;
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case SNDRV_PCM_FORMAT_S24_LE:
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case SNDRV_PCM_FORMAT_S32_LE:
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word_len = 0;
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break;
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}
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snd_soc_update_bits(codec, AD1836_DAC_CTRL1,
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AD1836_DAC_WORD_LEN_MASK, word_len);
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snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
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AD1836_ADC_WORD_LEN_MASK, word_len);
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return 0;
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}
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#ifdef CONFIG_PM
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static int ad1836_soc_suspend(struct platform_device *pdev,
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pm_message_t state)
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{
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struct snd_soc_device *socdev = platform_get_drvdata(pdev);
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struct snd_soc_codec *codec = socdev->card->codec;
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/* reset clock control mode */
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u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2);
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adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK;
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return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2);
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}
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static int ad1836_soc_resume(struct platform_device *pdev)
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{
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struct snd_soc_device *socdev = platform_get_drvdata(pdev);
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struct snd_soc_codec *codec = socdev->card->codec;
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/* restore clock control mode */
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u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2);
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adc_ctrl2 |= AD1836_ADC_AUX;
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return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2);
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}
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#else
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#define ad1836_soc_suspend NULL
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#define ad1836_soc_resume NULL
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#endif
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static int __devinit ad1836_spi_probe(struct spi_device *spi)
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{
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struct snd_soc_codec *codec;
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struct ad1836_priv *ad1836;
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ad1836 = kzalloc(sizeof(struct ad1836_priv), GFP_KERNEL);
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if (ad1836 == NULL)
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return -ENOMEM;
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codec = &ad1836->codec;
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codec->control_data = spi;
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codec->dev = &spi->dev;
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dev_set_drvdata(&spi->dev, ad1836);
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return ad1836_register(ad1836);
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}
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static int __devexit ad1836_spi_remove(struct spi_device *spi)
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{
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struct ad1836_priv *ad1836 = dev_get_drvdata(&spi->dev);
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ad1836_unregister(ad1836);
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return 0;
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}
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static struct spi_driver ad1836_spi_driver = {
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.driver = {
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.name = "ad1836",
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.owner = THIS_MODULE,
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},
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.probe = ad1836_spi_probe,
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.remove = __devexit_p(ad1836_spi_remove),
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};
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static struct snd_soc_dai_ops ad1836_dai_ops = {
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.hw_params = ad1836_hw_params,
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.set_fmt = ad1836_set_dai_fmt,
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};
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/* codec DAI instance */
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struct snd_soc_dai ad1836_dai = {
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.name = "AD1836",
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.playback = {
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.stream_name = "Playback",
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.channels_min = 2,
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.channels_max = 6,
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.rates = SNDRV_PCM_RATE_48000,
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.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
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SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
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},
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.capture = {
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.stream_name = "Capture",
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.channels_min = 2,
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.channels_max = 4,
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.rates = SNDRV_PCM_RATE_48000,
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.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
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SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
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},
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.ops = &ad1836_dai_ops,
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};
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EXPORT_SYMBOL_GPL(ad1836_dai);
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static int ad1836_register(struct ad1836_priv *ad1836)
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{
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int ret;
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struct snd_soc_codec *codec = &ad1836->codec;
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if (ad1836_codec) {
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dev_err(codec->dev, "Another ad1836 is registered\n");
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return -EINVAL;
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}
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mutex_init(&codec->mutex);
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INIT_LIST_HEAD(&codec->dapm_widgets);
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INIT_LIST_HEAD(&codec->dapm_paths);
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codec->private_data = ad1836;
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codec->reg_cache = ad1836->reg_cache;
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codec->reg_cache_size = AD1836_NUM_REGS;
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codec->name = "AD1836";
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codec->owner = THIS_MODULE;
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codec->dai = &ad1836_dai;
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codec->num_dai = 1;
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INIT_LIST_HEAD(&codec->dapm_widgets);
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INIT_LIST_HEAD(&codec->dapm_paths);
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ad1836_dai.dev = codec->dev;
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ad1836_codec = codec;
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ret = snd_soc_codec_set_cache_io(codec, 4, 12, SND_SOC_SPI);
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if (ret < 0) {
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dev_err(codec->dev, "failed to set cache I/O: %d\n",
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ret);
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kfree(ad1836);
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return ret;
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}
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/* default setting for ad1836 */
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/* de-emphasis: 48kHz, power-on dac */
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snd_soc_write(codec, AD1836_DAC_CTRL1, 0x300);
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/* unmute dac channels */
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snd_soc_write(codec, AD1836_DAC_CTRL2, 0x0);
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/* high-pass filter enable, power-on adc */
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snd_soc_write(codec, AD1836_ADC_CTRL1, 0x100);
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/* unmute adc channles, adc aux mode */
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snd_soc_write(codec, AD1836_ADC_CTRL2, 0x180);
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/* left/right diff:PGA/MUX */
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snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A);
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/* volume */
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snd_soc_write(codec, AD1836_DAC_L1_VOL, 0x3FF);
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snd_soc_write(codec, AD1836_DAC_R1_VOL, 0x3FF);
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snd_soc_write(codec, AD1836_DAC_L2_VOL, 0x3FF);
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snd_soc_write(codec, AD1836_DAC_R2_VOL, 0x3FF);
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snd_soc_write(codec, AD1836_DAC_L3_VOL, 0x3FF);
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snd_soc_write(codec, AD1836_DAC_R3_VOL, 0x3FF);
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ret = snd_soc_register_codec(codec);
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if (ret != 0) {
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dev_err(codec->dev, "Failed to register codec: %d\n", ret);
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kfree(ad1836);
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return ret;
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}
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ret = snd_soc_register_dai(&ad1836_dai);
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if (ret != 0) {
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dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
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snd_soc_unregister_codec(codec);
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kfree(ad1836);
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return ret;
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}
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return 0;
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}
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static void ad1836_unregister(struct ad1836_priv *ad1836)
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{
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snd_soc_unregister_dai(&ad1836_dai);
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snd_soc_unregister_codec(&ad1836->codec);
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kfree(ad1836);
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ad1836_codec = NULL;
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}
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static int ad1836_probe(struct platform_device *pdev)
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{
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struct snd_soc_device *socdev = platform_get_drvdata(pdev);
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struct snd_soc_codec *codec;
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int ret = 0;
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if (ad1836_codec == NULL) {
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dev_err(&pdev->dev, "Codec device not registered\n");
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return -ENODEV;
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}
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socdev->card->codec = ad1836_codec;
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codec = ad1836_codec;
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/* register pcms */
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ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
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if (ret < 0) {
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dev_err(codec->dev, "failed to create pcms: %d\n", ret);
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goto pcm_err;
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}
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snd_soc_add_controls(codec, ad1836_snd_controls,
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ARRAY_SIZE(ad1836_snd_controls));
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snd_soc_dapm_new_controls(codec, ad1836_dapm_widgets,
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ARRAY_SIZE(ad1836_dapm_widgets));
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snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
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pcm_err:
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return ret;
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}
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/* power down chip */
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static int ad1836_remove(struct platform_device *pdev)
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{
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struct snd_soc_device *socdev = platform_get_drvdata(pdev);
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snd_soc_free_pcms(socdev);
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snd_soc_dapm_free(socdev);
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return 0;
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}
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struct snd_soc_codec_device soc_codec_dev_ad1836 = {
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.probe = ad1836_probe,
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.remove = ad1836_remove,
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.suspend = ad1836_soc_suspend,
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.resume = ad1836_soc_resume,
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};
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EXPORT_SYMBOL_GPL(soc_codec_dev_ad1836);
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static int __init ad1836_init(void)
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{
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int ret;
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ret = spi_register_driver(&ad1836_spi_driver);
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if (ret != 0) {
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printk(KERN_ERR "Failed to register ad1836 SPI driver: %d\n",
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ret);
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}
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return ret;
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}
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module_init(ad1836_init);
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static void __exit ad1836_exit(void)
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{
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spi_unregister_driver(&ad1836_spi_driver);
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}
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module_exit(ad1836_exit);
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MODULE_DESCRIPTION("ASoC ad1836 driver");
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MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>");
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MODULE_LICENSE("GPL");
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