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faa38b5e0e
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (214 commits) ALSA: hda - Add pin-fix for HP dc5750 ALSA: als4000: Fix potentially invalid DMA mode setup ALSA: als4000: enable burst mode ALSA: hda - Fix initial capsrc selection in patch_alc269() ASoC: TWL4030: Capture route runtime DAPM ordering fix ALSA: hda - Add PC-beep whitelist for an Intel board ALSA: hda - More relax for pending period handling ALSA: hda - Define AC_FMT_* constants ALSA: hda - Fix beep frequency on IDT 92HD73xx and 92HD71Bxx codecs ALSA: hda - Add support for HDMI HBR passthrough ALSA: hda - Set Stream Type in Stream Format according to AES0 ALSA: hda - Fix Thinkpad X300 so SPDIF is not exposed ALSA: hda - FIX to not expose SPDIF on Thinkpad X301, since it does not have the ability to use SPDIF ASoC: wm9081: fix resource reclaim in wm9081_register error path ASoC: wm8978: fix a memory leak if a wm8978_register fail ASoC: wm8974: fix a memory leak if another WM8974 is registered ASoC: wm8961: fix resource reclaim in wm8961_register error path ASoC: wm8955: fix resource reclaim in wm8955_register error path ASoC: wm8940: fix a memory leak if wm8940_register return error ASoC: wm8904: fix resource reclaim in wm8904_register error path ...
2897 lines
76 KiB
C
2897 lines
76 KiB
C
/*
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* soc-core.c -- ALSA SoC Audio Layer
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*
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* Copyright 2005 Wolfson Microelectronics PLC.
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* Copyright 2005 Openedhand Ltd.
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*
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* Author: Liam Girdwood <lrg@slimlogic.co.uk>
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* with code, comments and ideas from :-
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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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* TODO:
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* o Add hw rules to enforce rates, etc.
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* o More testing with other codecs/machines.
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* o Add more codecs and platforms to ensure good API coverage.
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* o Support TDM on PCM and I2S
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*/
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#include <linux/module.h>
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#include <linux/moduleparam.h>
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#include <linux/init.h>
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#include <linux/delay.h>
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#include <linux/pm.h>
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#include <linux/bitops.h>
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#include <linux/debugfs.h>
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#include <linux/platform_device.h>
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#include <linux/slab.h>
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#include <sound/ac97_codec.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/initval.h>
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static DEFINE_MUTEX(pcm_mutex);
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static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
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#ifdef CONFIG_DEBUG_FS
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static struct dentry *debugfs_root;
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#endif
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static DEFINE_MUTEX(client_mutex);
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static LIST_HEAD(card_list);
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static LIST_HEAD(dai_list);
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static LIST_HEAD(platform_list);
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static LIST_HEAD(codec_list);
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static int snd_soc_register_card(struct snd_soc_card *card);
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static int snd_soc_unregister_card(struct snd_soc_card *card);
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/*
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* This is a timeout to do a DAPM powerdown after a stream is closed().
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* It can be used to eliminate pops between different playback streams, e.g.
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* between two audio tracks.
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*/
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static int pmdown_time = 5000;
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module_param(pmdown_time, int, 0);
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MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
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/*
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* This function forces any delayed work to be queued and run.
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*/
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static int run_delayed_work(struct delayed_work *dwork)
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{
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int ret;
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/* cancel any work waiting to be queued. */
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ret = cancel_delayed_work(dwork);
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/* if there was any work waiting then we run it now and
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* wait for it's completion */
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if (ret) {
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schedule_delayed_work(dwork, 0);
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flush_scheduled_work();
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}
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return ret;
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}
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/* codec register dump */
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static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
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{
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int ret, i, step = 1, count = 0;
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if (!codec->reg_cache_size)
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return 0;
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if (codec->reg_cache_step)
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step = codec->reg_cache_step;
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count += sprintf(buf, "%s registers\n", codec->name);
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for (i = 0; i < codec->reg_cache_size; i += step) {
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if (codec->readable_register && !codec->readable_register(i))
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continue;
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count += sprintf(buf + count, "%2x: ", i);
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if (count >= PAGE_SIZE - 1)
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break;
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if (codec->display_register) {
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count += codec->display_register(codec, buf + count,
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PAGE_SIZE - count, i);
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} else {
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/* If the read fails it's almost certainly due to
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* the register being volatile and the device being
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* powered off.
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*/
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ret = codec->read(codec, i);
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if (ret >= 0)
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count += snprintf(buf + count,
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PAGE_SIZE - count,
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"%4x", ret);
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else
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count += snprintf(buf + count,
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PAGE_SIZE - count,
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"<no data: %d>", ret);
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}
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if (count >= PAGE_SIZE - 1)
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break;
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count += snprintf(buf + count, PAGE_SIZE - count, "\n");
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if (count >= PAGE_SIZE - 1)
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break;
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}
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/* Truncate count; min() would cause a warning */
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if (count >= PAGE_SIZE)
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count = PAGE_SIZE - 1;
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return count;
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}
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static ssize_t codec_reg_show(struct device *dev,
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struct device_attribute *attr, char *buf)
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{
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struct snd_soc_device *devdata = dev_get_drvdata(dev);
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return soc_codec_reg_show(devdata->card->codec, buf);
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}
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static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
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static ssize_t pmdown_time_show(struct device *dev,
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struct device_attribute *attr, char *buf)
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{
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struct snd_soc_device *socdev = dev_get_drvdata(dev);
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struct snd_soc_card *card = socdev->card;
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return sprintf(buf, "%ld\n", card->pmdown_time);
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}
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static ssize_t pmdown_time_set(struct device *dev,
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struct device_attribute *attr,
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const char *buf, size_t count)
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{
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struct snd_soc_device *socdev = dev_get_drvdata(dev);
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struct snd_soc_card *card = socdev->card;
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strict_strtol(buf, 10, &card->pmdown_time);
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return count;
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}
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static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set);
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#ifdef CONFIG_DEBUG_FS
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static int codec_reg_open_file(struct inode *inode, struct file *file)
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{
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file->private_data = inode->i_private;
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return 0;
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}
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static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf,
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size_t count, loff_t *ppos)
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{
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ssize_t ret;
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struct snd_soc_codec *codec = file->private_data;
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char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
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if (!buf)
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return -ENOMEM;
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ret = soc_codec_reg_show(codec, buf);
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if (ret >= 0)
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ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
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kfree(buf);
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return ret;
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}
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static ssize_t codec_reg_write_file(struct file *file,
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const char __user *user_buf, size_t count, loff_t *ppos)
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{
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char buf[32];
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int buf_size;
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char *start = buf;
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unsigned long reg, value;
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int step = 1;
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struct snd_soc_codec *codec = file->private_data;
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buf_size = min(count, (sizeof(buf)-1));
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if (copy_from_user(buf, user_buf, buf_size))
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return -EFAULT;
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buf[buf_size] = 0;
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if (codec->reg_cache_step)
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step = codec->reg_cache_step;
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while (*start == ' ')
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start++;
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reg = simple_strtoul(start, &start, 16);
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if ((reg >= codec->reg_cache_size) || (reg % step))
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return -EINVAL;
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while (*start == ' ')
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start++;
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if (strict_strtoul(start, 16, &value))
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return -EINVAL;
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codec->write(codec, reg, value);
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return buf_size;
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}
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static const struct file_operations codec_reg_fops = {
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.open = codec_reg_open_file,
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.read = codec_reg_read_file,
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.write = codec_reg_write_file,
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};
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static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
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{
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char codec_root[128];
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if (codec->dev)
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snprintf(codec_root, sizeof(codec_root),
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"%s.%s", codec->name, dev_name(codec->dev));
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else
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snprintf(codec_root, sizeof(codec_root),
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"%s", codec->name);
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codec->debugfs_codec_root = debugfs_create_dir(codec_root,
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debugfs_root);
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if (!codec->debugfs_codec_root) {
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printk(KERN_WARNING
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"ASoC: Failed to create codec debugfs directory\n");
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return;
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}
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codec->debugfs_reg = debugfs_create_file("codec_reg", 0644,
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codec->debugfs_codec_root,
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codec, &codec_reg_fops);
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if (!codec->debugfs_reg)
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printk(KERN_WARNING
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"ASoC: Failed to create codec register debugfs file\n");
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codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744,
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codec->debugfs_codec_root,
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&codec->pop_time);
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if (!codec->debugfs_pop_time)
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printk(KERN_WARNING
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"Failed to create pop time debugfs file\n");
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codec->debugfs_dapm = debugfs_create_dir("dapm",
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codec->debugfs_codec_root);
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if (!codec->debugfs_dapm)
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printk(KERN_WARNING
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"Failed to create DAPM debugfs directory\n");
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snd_soc_dapm_debugfs_init(codec);
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}
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static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
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{
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debugfs_remove_recursive(codec->debugfs_codec_root);
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}
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#else
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static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec)
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{
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}
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static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
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{
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}
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#endif
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#ifdef CONFIG_SND_SOC_AC97_BUS
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/* unregister ac97 codec */
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static int soc_ac97_dev_unregister(struct snd_soc_codec *codec)
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{
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if (codec->ac97->dev.bus)
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device_unregister(&codec->ac97->dev);
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return 0;
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}
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/* stop no dev release warning */
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static void soc_ac97_device_release(struct device *dev){}
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/* register ac97 codec to bus */
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static int soc_ac97_dev_register(struct snd_soc_codec *codec)
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{
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int err;
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codec->ac97->dev.bus = &ac97_bus_type;
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codec->ac97->dev.parent = codec->card->dev;
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codec->ac97->dev.release = soc_ac97_device_release;
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dev_set_name(&codec->ac97->dev, "%d-%d:%s",
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codec->card->number, 0, codec->name);
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err = device_register(&codec->ac97->dev);
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if (err < 0) {
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snd_printk(KERN_ERR "Can't register ac97 bus\n");
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codec->ac97->dev.bus = NULL;
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return err;
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}
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return 0;
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}
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#endif
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static int soc_pcm_apply_symmetry(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_device *socdev = rtd->socdev;
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struct snd_soc_card *card = socdev->card;
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struct snd_soc_dai_link *machine = rtd->dai;
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struct snd_soc_dai *cpu_dai = machine->cpu_dai;
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struct snd_soc_dai *codec_dai = machine->codec_dai;
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int ret;
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if (codec_dai->symmetric_rates || cpu_dai->symmetric_rates ||
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machine->symmetric_rates) {
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dev_dbg(card->dev, "Symmetry forces %dHz rate\n",
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machine->rate);
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ret = snd_pcm_hw_constraint_minmax(substream->runtime,
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SNDRV_PCM_HW_PARAM_RATE,
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machine->rate,
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machine->rate);
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if (ret < 0) {
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dev_err(card->dev,
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"Unable to apply rate symmetry constraint: %d\n", ret);
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return ret;
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}
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}
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return 0;
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}
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/*
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* Called by ALSA when a PCM substream is opened, the runtime->hw record is
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* then initialized and any private data can be allocated. This also calls
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* startup for the cpu DAI, platform, machine and codec DAI.
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*/
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static int soc_pcm_open(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_device *socdev = rtd->socdev;
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struct snd_soc_card *card = socdev->card;
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struct snd_pcm_runtime *runtime = substream->runtime;
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struct snd_soc_dai_link *machine = rtd->dai;
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struct snd_soc_platform *platform = card->platform;
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struct snd_soc_dai *cpu_dai = machine->cpu_dai;
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struct snd_soc_dai *codec_dai = machine->codec_dai;
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int ret = 0;
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mutex_lock(&pcm_mutex);
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|
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/* startup the audio subsystem */
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if (cpu_dai->ops->startup) {
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ret = cpu_dai->ops->startup(substream, cpu_dai);
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if (ret < 0) {
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printk(KERN_ERR "asoc: can't open interface %s\n",
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cpu_dai->name);
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goto out;
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}
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}
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|
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if (platform->pcm_ops->open) {
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ret = platform->pcm_ops->open(substream);
|
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if (ret < 0) {
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printk(KERN_ERR "asoc: can't open platform %s\n", platform->name);
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goto platform_err;
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}
|
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}
|
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|
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if (codec_dai->ops->startup) {
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ret = codec_dai->ops->startup(substream, codec_dai);
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if (ret < 0) {
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printk(KERN_ERR "asoc: can't open codec %s\n",
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codec_dai->name);
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goto codec_dai_err;
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}
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}
|
|
|
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if (machine->ops && machine->ops->startup) {
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ret = machine->ops->startup(substream);
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if (ret < 0) {
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printk(KERN_ERR "asoc: %s startup failed\n", machine->name);
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goto machine_err;
|
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}
|
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}
|
|
|
|
/* Check that the codec and cpu DAI's are compatible */
|
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
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runtime->hw.rate_min =
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max(codec_dai->playback.rate_min,
|
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cpu_dai->playback.rate_min);
|
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runtime->hw.rate_max =
|
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min(codec_dai->playback.rate_max,
|
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cpu_dai->playback.rate_max);
|
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runtime->hw.channels_min =
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max(codec_dai->playback.channels_min,
|
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cpu_dai->playback.channels_min);
|
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runtime->hw.channels_max =
|
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min(codec_dai->playback.channels_max,
|
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cpu_dai->playback.channels_max);
|
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runtime->hw.formats =
|
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codec_dai->playback.formats & cpu_dai->playback.formats;
|
|
runtime->hw.rates =
|
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codec_dai->playback.rates & cpu_dai->playback.rates;
|
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if (codec_dai->playback.rates
|
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& (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
|
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runtime->hw.rates |= cpu_dai->playback.rates;
|
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if (cpu_dai->playback.rates
|
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& (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
|
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runtime->hw.rates |= codec_dai->playback.rates;
|
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} else {
|
|
runtime->hw.rate_min =
|
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max(codec_dai->capture.rate_min,
|
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cpu_dai->capture.rate_min);
|
|
runtime->hw.rate_max =
|
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min(codec_dai->capture.rate_max,
|
|
cpu_dai->capture.rate_max);
|
|
runtime->hw.channels_min =
|
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max(codec_dai->capture.channels_min,
|
|
cpu_dai->capture.channels_min);
|
|
runtime->hw.channels_max =
|
|
min(codec_dai->capture.channels_max,
|
|
cpu_dai->capture.channels_max);
|
|
runtime->hw.formats =
|
|
codec_dai->capture.formats & cpu_dai->capture.formats;
|
|
runtime->hw.rates =
|
|
codec_dai->capture.rates & cpu_dai->capture.rates;
|
|
if (codec_dai->capture.rates
|
|
& (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
|
|
runtime->hw.rates |= cpu_dai->capture.rates;
|
|
if (cpu_dai->capture.rates
|
|
& (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
|
|
runtime->hw.rates |= codec_dai->capture.rates;
|
|
}
|
|
|
|
snd_pcm_limit_hw_rates(runtime);
|
|
if (!runtime->hw.rates) {
|
|
printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
|
|
codec_dai->name, cpu_dai->name);
|
|
goto config_err;
|
|
}
|
|
if (!runtime->hw.formats) {
|
|
printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
|
|
codec_dai->name, cpu_dai->name);
|
|
goto config_err;
|
|
}
|
|
if (!runtime->hw.channels_min || !runtime->hw.channels_max) {
|
|
printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
|
|
codec_dai->name, cpu_dai->name);
|
|
goto config_err;
|
|
}
|
|
|
|
/* Symmetry only applies if we've already got an active stream. */
|
|
if (cpu_dai->active || codec_dai->active) {
|
|
ret = soc_pcm_apply_symmetry(substream);
|
|
if (ret != 0)
|
|
goto config_err;
|
|
}
|
|
|
|
pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name);
|
|
pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates);
|
|
pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
|
|
runtime->hw.channels_max);
|
|
pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
|
|
runtime->hw.rate_max);
|
|
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
|
|
cpu_dai->playback.active++;
|
|
codec_dai->playback.active++;
|
|
} else {
|
|
cpu_dai->capture.active++;
|
|
codec_dai->capture.active++;
|
|
}
|
|
cpu_dai->active++;
|
|
codec_dai->active++;
|
|
card->codec->active++;
|
|
mutex_unlock(&pcm_mutex);
|
|
return 0;
|
|
|
|
config_err:
|
|
if (machine->ops && machine->ops->shutdown)
|
|
machine->ops->shutdown(substream);
|
|
|
|
machine_err:
|
|
if (codec_dai->ops->shutdown)
|
|
codec_dai->ops->shutdown(substream, codec_dai);
|
|
|
|
codec_dai_err:
|
|
if (platform->pcm_ops->close)
|
|
platform->pcm_ops->close(substream);
|
|
|
|
platform_err:
|
|
if (cpu_dai->ops->shutdown)
|
|
cpu_dai->ops->shutdown(substream, cpu_dai);
|
|
out:
|
|
mutex_unlock(&pcm_mutex);
|
|
return ret;
|
|
}
|
|
|
|
/*
|
|
* Power down the audio subsystem pmdown_time msecs after close is called.
|
|
* This is to ensure there are no pops or clicks in between any music tracks
|
|
* due to DAPM power cycling.
|
|
*/
|
|
static void close_delayed_work(struct work_struct *work)
|
|
{
|
|
struct snd_soc_card *card = container_of(work, struct snd_soc_card,
|
|
delayed_work.work);
|
|
struct snd_soc_codec *codec = card->codec;
|
|
struct snd_soc_dai *codec_dai;
|
|
int i;
|
|
|
|
mutex_lock(&pcm_mutex);
|
|
for (i = 0; i < codec->num_dai; i++) {
|
|
codec_dai = &codec->dai[i];
|
|
|
|
pr_debug("pop wq checking: %s status: %s waiting: %s\n",
|
|
codec_dai->playback.stream_name,
|
|
codec_dai->playback.active ? "active" : "inactive",
|
|
codec_dai->pop_wait ? "yes" : "no");
|
|
|
|
/* are we waiting on this codec DAI stream */
|
|
if (codec_dai->pop_wait == 1) {
|
|
codec_dai->pop_wait = 0;
|
|
snd_soc_dapm_stream_event(codec,
|
|
codec_dai->playback.stream_name,
|
|
SND_SOC_DAPM_STREAM_STOP);
|
|
}
|
|
}
|
|
mutex_unlock(&pcm_mutex);
|
|
}
|
|
|
|
/*
|
|
* Called by ALSA when a PCM substream is closed. Private data can be
|
|
* freed here. The cpu DAI, codec DAI, machine and platform are also
|
|
* shutdown.
|
|
*/
|
|
static int soc_codec_close(struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_soc_pcm_runtime *rtd = substream->private_data;
|
|
struct snd_soc_device *socdev = rtd->socdev;
|
|
struct snd_soc_card *card = socdev->card;
|
|
struct snd_soc_dai_link *machine = rtd->dai;
|
|
struct snd_soc_platform *platform = card->platform;
|
|
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
|
|
struct snd_soc_dai *codec_dai = machine->codec_dai;
|
|
struct snd_soc_codec *codec = card->codec;
|
|
|
|
mutex_lock(&pcm_mutex);
|
|
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
|
|
cpu_dai->playback.active--;
|
|
codec_dai->playback.active--;
|
|
} else {
|
|
cpu_dai->capture.active--;
|
|
codec_dai->capture.active--;
|
|
}
|
|
|
|
cpu_dai->active--;
|
|
codec_dai->active--;
|
|
codec->active--;
|
|
|
|
/* Muting the DAC suppresses artifacts caused during digital
|
|
* shutdown, for example from stopping clocks.
|
|
*/
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
|
|
snd_soc_dai_digital_mute(codec_dai, 1);
|
|
|
|
if (cpu_dai->ops->shutdown)
|
|
cpu_dai->ops->shutdown(substream, cpu_dai);
|
|
|
|
if (codec_dai->ops->shutdown)
|
|
codec_dai->ops->shutdown(substream, codec_dai);
|
|
|
|
if (machine->ops && machine->ops->shutdown)
|
|
machine->ops->shutdown(substream);
|
|
|
|
if (platform->pcm_ops->close)
|
|
platform->pcm_ops->close(substream);
|
|
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
|
|
/* start delayed pop wq here for playback streams */
|
|
codec_dai->pop_wait = 1;
|
|
schedule_delayed_work(&card->delayed_work,
|
|
msecs_to_jiffies(card->pmdown_time));
|
|
} else {
|
|
/* capture streams can be powered down now */
|
|
snd_soc_dapm_stream_event(codec,
|
|
codec_dai->capture.stream_name,
|
|
SND_SOC_DAPM_STREAM_STOP);
|
|
}
|
|
|
|
mutex_unlock(&pcm_mutex);
|
|
return 0;
|
|
}
|
|
|
|
/*
|
|
* Called by ALSA when the PCM substream is prepared, can set format, sample
|
|
* rate, etc. This function is non atomic and can be called multiple times,
|
|
* it can refer to the runtime info.
|
|
*/
|
|
static int soc_pcm_prepare(struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_soc_pcm_runtime *rtd = substream->private_data;
|
|
struct snd_soc_device *socdev = rtd->socdev;
|
|
struct snd_soc_card *card = socdev->card;
|
|
struct snd_soc_dai_link *machine = rtd->dai;
|
|
struct snd_soc_platform *platform = card->platform;
|
|
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
|
|
struct snd_soc_dai *codec_dai = machine->codec_dai;
|
|
struct snd_soc_codec *codec = card->codec;
|
|
int ret = 0;
|
|
|
|
mutex_lock(&pcm_mutex);
|
|
|
|
if (machine->ops && machine->ops->prepare) {
|
|
ret = machine->ops->prepare(substream);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: machine prepare error\n");
|
|
goto out;
|
|
}
|
|
}
|
|
|
|
if (platform->pcm_ops->prepare) {
|
|
ret = platform->pcm_ops->prepare(substream);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: platform prepare error\n");
|
|
goto out;
|
|
}
|
|
}
|
|
|
|
if (codec_dai->ops->prepare) {
|
|
ret = codec_dai->ops->prepare(substream, codec_dai);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: codec DAI prepare error\n");
|
|
goto out;
|
|
}
|
|
}
|
|
|
|
if (cpu_dai->ops->prepare) {
|
|
ret = cpu_dai->ops->prepare(substream, cpu_dai);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: cpu DAI prepare error\n");
|
|
goto out;
|
|
}
|
|
}
|
|
|
|
/* cancel any delayed stream shutdown that is pending */
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
|
|
codec_dai->pop_wait) {
|
|
codec_dai->pop_wait = 0;
|
|
cancel_delayed_work(&card->delayed_work);
|
|
}
|
|
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
|
|
snd_soc_dapm_stream_event(codec,
|
|
codec_dai->playback.stream_name,
|
|
SND_SOC_DAPM_STREAM_START);
|
|
else
|
|
snd_soc_dapm_stream_event(codec,
|
|
codec_dai->capture.stream_name,
|
|
SND_SOC_DAPM_STREAM_START);
|
|
|
|
snd_soc_dai_digital_mute(codec_dai, 0);
|
|
|
|
out:
|
|
mutex_unlock(&pcm_mutex);
|
|
return ret;
|
|
}
|
|
|
|
/*
|
|
* Called by ALSA when the hardware params are set by application. This
|
|
* function can also be called multiple times and can allocate buffers
|
|
* (using snd_pcm_lib_* ). It's non-atomic.
|
|
*/
|
|
static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
|
|
struct snd_pcm_hw_params *params)
|
|
{
|
|
struct snd_soc_pcm_runtime *rtd = substream->private_data;
|
|
struct snd_soc_device *socdev = rtd->socdev;
|
|
struct snd_soc_dai_link *machine = rtd->dai;
|
|
struct snd_soc_card *card = socdev->card;
|
|
struct snd_soc_platform *platform = card->platform;
|
|
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
|
|
struct snd_soc_dai *codec_dai = machine->codec_dai;
|
|
int ret = 0;
|
|
|
|
mutex_lock(&pcm_mutex);
|
|
|
|
if (machine->ops && machine->ops->hw_params) {
|
|
ret = machine->ops->hw_params(substream, params);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: machine hw_params failed\n");
|
|
goto out;
|
|
}
|
|
}
|
|
|
|
if (codec_dai->ops->hw_params) {
|
|
ret = codec_dai->ops->hw_params(substream, params, codec_dai);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: can't set codec %s hw params\n",
|
|
codec_dai->name);
|
|
goto codec_err;
|
|
}
|
|
}
|
|
|
|
if (cpu_dai->ops->hw_params) {
|
|
ret = cpu_dai->ops->hw_params(substream, params, cpu_dai);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: interface %s hw params failed\n",
|
|
cpu_dai->name);
|
|
goto interface_err;
|
|
}
|
|
}
|
|
|
|
if (platform->pcm_ops->hw_params) {
|
|
ret = platform->pcm_ops->hw_params(substream, params);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: platform %s hw params failed\n",
|
|
platform->name);
|
|
goto platform_err;
|
|
}
|
|
}
|
|
|
|
machine->rate = params_rate(params);
|
|
|
|
out:
|
|
mutex_unlock(&pcm_mutex);
|
|
return ret;
|
|
|
|
platform_err:
|
|
if (cpu_dai->ops->hw_free)
|
|
cpu_dai->ops->hw_free(substream, cpu_dai);
|
|
|
|
interface_err:
|
|
if (codec_dai->ops->hw_free)
|
|
codec_dai->ops->hw_free(substream, codec_dai);
|
|
|
|
codec_err:
|
|
if (machine->ops && machine->ops->hw_free)
|
|
machine->ops->hw_free(substream);
|
|
|
|
mutex_unlock(&pcm_mutex);
|
|
return ret;
|
|
}
|
|
|
|
/*
|
|
* Free's resources allocated by hw_params, can be called multiple times
|
|
*/
|
|
static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_soc_pcm_runtime *rtd = substream->private_data;
|
|
struct snd_soc_device *socdev = rtd->socdev;
|
|
struct snd_soc_dai_link *machine = rtd->dai;
|
|
struct snd_soc_card *card = socdev->card;
|
|
struct snd_soc_platform *platform = card->platform;
|
|
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
|
|
struct snd_soc_dai *codec_dai = machine->codec_dai;
|
|
struct snd_soc_codec *codec = card->codec;
|
|
|
|
mutex_lock(&pcm_mutex);
|
|
|
|
/* apply codec digital mute */
|
|
if (!codec->active)
|
|
snd_soc_dai_digital_mute(codec_dai, 1);
|
|
|
|
/* free any machine hw params */
|
|
if (machine->ops && machine->ops->hw_free)
|
|
machine->ops->hw_free(substream);
|
|
|
|
/* free any DMA resources */
|
|
if (platform->pcm_ops->hw_free)
|
|
platform->pcm_ops->hw_free(substream);
|
|
|
|
/* now free hw params for the DAI's */
|
|
if (codec_dai->ops->hw_free)
|
|
codec_dai->ops->hw_free(substream, codec_dai);
|
|
|
|
if (cpu_dai->ops->hw_free)
|
|
cpu_dai->ops->hw_free(substream, cpu_dai);
|
|
|
|
mutex_unlock(&pcm_mutex);
|
|
return 0;
|
|
}
|
|
|
|
static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
|
|
{
|
|
struct snd_soc_pcm_runtime *rtd = substream->private_data;
|
|
struct snd_soc_device *socdev = rtd->socdev;
|
|
struct snd_soc_card *card= socdev->card;
|
|
struct snd_soc_dai_link *machine = rtd->dai;
|
|
struct snd_soc_platform *platform = card->platform;
|
|
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
|
|
struct snd_soc_dai *codec_dai = machine->codec_dai;
|
|
int ret;
|
|
|
|
if (codec_dai->ops->trigger) {
|
|
ret = codec_dai->ops->trigger(substream, cmd, codec_dai);
|
|
if (ret < 0)
|
|
return ret;
|
|
}
|
|
|
|
if (platform->pcm_ops->trigger) {
|
|
ret = platform->pcm_ops->trigger(substream, cmd);
|
|
if (ret < 0)
|
|
return ret;
|
|
}
|
|
|
|
if (cpu_dai->ops->trigger) {
|
|
ret = cpu_dai->ops->trigger(substream, cmd, cpu_dai);
|
|
if (ret < 0)
|
|
return ret;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/*
|
|
* soc level wrapper for pointer callback
|
|
* If cpu_dai, codec_dai, platform driver has the delay callback, than
|
|
* the runtime->delay will be updated accordingly.
|
|
*/
|
|
static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_soc_pcm_runtime *rtd = substream->private_data;
|
|
struct snd_soc_device *socdev = rtd->socdev;
|
|
struct snd_soc_card *card = socdev->card;
|
|
struct snd_soc_platform *platform = card->platform;
|
|
struct snd_soc_dai_link *machine = rtd->dai;
|
|
struct snd_soc_dai *cpu_dai = machine->cpu_dai;
|
|
struct snd_soc_dai *codec_dai = machine->codec_dai;
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
snd_pcm_uframes_t offset = 0;
|
|
snd_pcm_sframes_t delay = 0;
|
|
|
|
if (platform->pcm_ops->pointer)
|
|
offset = platform->pcm_ops->pointer(substream);
|
|
|
|
if (cpu_dai->ops->delay)
|
|
delay += cpu_dai->ops->delay(substream, cpu_dai);
|
|
|
|
if (codec_dai->ops->delay)
|
|
delay += codec_dai->ops->delay(substream, codec_dai);
|
|
|
|
if (platform->delay)
|
|
delay += platform->delay(substream, codec_dai);
|
|
|
|
runtime->delay = delay;
|
|
|
|
return offset;
|
|
}
|
|
|
|
/* ASoC PCM operations */
|
|
static struct snd_pcm_ops soc_pcm_ops = {
|
|
.open = soc_pcm_open,
|
|
.close = soc_codec_close,
|
|
.hw_params = soc_pcm_hw_params,
|
|
.hw_free = soc_pcm_hw_free,
|
|
.prepare = soc_pcm_prepare,
|
|
.trigger = soc_pcm_trigger,
|
|
.pointer = soc_pcm_pointer,
|
|
};
|
|
|
|
#ifdef CONFIG_PM
|
|
/* powers down audio subsystem for suspend */
|
|
static int soc_suspend(struct device *dev)
|
|
{
|
|
struct platform_device *pdev = to_platform_device(dev);
|
|
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
|
|
struct snd_soc_card *card = socdev->card;
|
|
struct snd_soc_platform *platform = card->platform;
|
|
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
|
|
struct snd_soc_codec *codec = card->codec;
|
|
int i;
|
|
|
|
/* If the initialization of this soc device failed, there is no codec
|
|
* associated with it. Just bail out in this case.
|
|
*/
|
|
if (!codec)
|
|
return 0;
|
|
|
|
/* Due to the resume being scheduled into a workqueue we could
|
|
* suspend before that's finished - wait for it to complete.
|
|
*/
|
|
snd_power_lock(codec->card);
|
|
snd_power_wait(codec->card, SNDRV_CTL_POWER_D0);
|
|
snd_power_unlock(codec->card);
|
|
|
|
/* we're going to block userspace touching us until resume completes */
|
|
snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot);
|
|
|
|
/* mute any active DAC's */
|
|
for (i = 0; i < card->num_links; i++) {
|
|
struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
|
|
|
|
if (card->dai_link[i].ignore_suspend)
|
|
continue;
|
|
|
|
if (dai->ops->digital_mute && dai->playback.active)
|
|
dai->ops->digital_mute(dai, 1);
|
|
}
|
|
|
|
/* suspend all pcms */
|
|
for (i = 0; i < card->num_links; i++) {
|
|
if (card->dai_link[i].ignore_suspend)
|
|
continue;
|
|
|
|
snd_pcm_suspend_all(card->dai_link[i].pcm);
|
|
}
|
|
|
|
if (card->suspend_pre)
|
|
card->suspend_pre(pdev, PMSG_SUSPEND);
|
|
|
|
for (i = 0; i < card->num_links; i++) {
|
|
struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
|
|
|
|
if (card->dai_link[i].ignore_suspend)
|
|
continue;
|
|
|
|
if (cpu_dai->suspend && !cpu_dai->ac97_control)
|
|
cpu_dai->suspend(cpu_dai);
|
|
if (platform->suspend)
|
|
platform->suspend(&card->dai_link[i]);
|
|
}
|
|
|
|
/* close any waiting streams and save state */
|
|
run_delayed_work(&card->delayed_work);
|
|
codec->suspend_bias_level = codec->bias_level;
|
|
|
|
for (i = 0; i < codec->num_dai; i++) {
|
|
char *stream = codec->dai[i].playback.stream_name;
|
|
|
|
if (card->dai_link[i].ignore_suspend)
|
|
continue;
|
|
|
|
if (stream != NULL)
|
|
snd_soc_dapm_stream_event(codec, stream,
|
|
SND_SOC_DAPM_STREAM_SUSPEND);
|
|
stream = codec->dai[i].capture.stream_name;
|
|
if (stream != NULL)
|
|
snd_soc_dapm_stream_event(codec, stream,
|
|
SND_SOC_DAPM_STREAM_SUSPEND);
|
|
}
|
|
|
|
/* If there are paths active then the CODEC will be held with
|
|
* bias _ON and should not be suspended. */
|
|
if (codec_dev->suspend) {
|
|
switch (codec->bias_level) {
|
|
case SND_SOC_BIAS_STANDBY:
|
|
case SND_SOC_BIAS_OFF:
|
|
codec_dev->suspend(pdev, PMSG_SUSPEND);
|
|
break;
|
|
default:
|
|
dev_dbg(socdev->dev, "CODEC is on over suspend\n");
|
|
break;
|
|
}
|
|
}
|
|
|
|
for (i = 0; i < card->num_links; i++) {
|
|
struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
|
|
|
|
if (card->dai_link[i].ignore_suspend)
|
|
continue;
|
|
|
|
if (cpu_dai->suspend && cpu_dai->ac97_control)
|
|
cpu_dai->suspend(cpu_dai);
|
|
}
|
|
|
|
if (card->suspend_post)
|
|
card->suspend_post(pdev, PMSG_SUSPEND);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* deferred resume work, so resume can complete before we finished
|
|
* setting our codec back up, which can be very slow on I2C
|
|
*/
|
|
static void soc_resume_deferred(struct work_struct *work)
|
|
{
|
|
struct snd_soc_card *card = container_of(work,
|
|
struct snd_soc_card,
|
|
deferred_resume_work);
|
|
struct snd_soc_device *socdev = card->socdev;
|
|
struct snd_soc_platform *platform = card->platform;
|
|
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
|
|
struct snd_soc_codec *codec = card->codec;
|
|
struct platform_device *pdev = to_platform_device(socdev->dev);
|
|
int i;
|
|
|
|
/* our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
|
|
* so userspace apps are blocked from touching us
|
|
*/
|
|
|
|
dev_dbg(socdev->dev, "starting resume work\n");
|
|
|
|
/* Bring us up into D2 so that DAPM starts enabling things */
|
|
snd_power_change_state(codec->card, SNDRV_CTL_POWER_D2);
|
|
|
|
if (card->resume_pre)
|
|
card->resume_pre(pdev);
|
|
|
|
for (i = 0; i < card->num_links; i++) {
|
|
struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
|
|
|
|
if (card->dai_link[i].ignore_suspend)
|
|
continue;
|
|
|
|
if (cpu_dai->resume && cpu_dai->ac97_control)
|
|
cpu_dai->resume(cpu_dai);
|
|
}
|
|
|
|
/* If the CODEC was idle over suspend then it will have been
|
|
* left with bias OFF or STANDBY and suspended so we must now
|
|
* resume. Otherwise the suspend was suppressed.
|
|
*/
|
|
if (codec_dev->resume) {
|
|
switch (codec->bias_level) {
|
|
case SND_SOC_BIAS_STANDBY:
|
|
case SND_SOC_BIAS_OFF:
|
|
codec_dev->resume(pdev);
|
|
break;
|
|
default:
|
|
dev_dbg(socdev->dev, "CODEC was on over suspend\n");
|
|
break;
|
|
}
|
|
}
|
|
|
|
for (i = 0; i < codec->num_dai; i++) {
|
|
char *stream = codec->dai[i].playback.stream_name;
|
|
|
|
if (card->dai_link[i].ignore_suspend)
|
|
continue;
|
|
|
|
if (stream != NULL)
|
|
snd_soc_dapm_stream_event(codec, stream,
|
|
SND_SOC_DAPM_STREAM_RESUME);
|
|
stream = codec->dai[i].capture.stream_name;
|
|
if (stream != NULL)
|
|
snd_soc_dapm_stream_event(codec, stream,
|
|
SND_SOC_DAPM_STREAM_RESUME);
|
|
}
|
|
|
|
/* unmute any active DACs */
|
|
for (i = 0; i < card->num_links; i++) {
|
|
struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
|
|
|
|
if (card->dai_link[i].ignore_suspend)
|
|
continue;
|
|
|
|
if (dai->ops->digital_mute && dai->playback.active)
|
|
dai->ops->digital_mute(dai, 0);
|
|
}
|
|
|
|
for (i = 0; i < card->num_links; i++) {
|
|
struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
|
|
|
|
if (card->dai_link[i].ignore_suspend)
|
|
continue;
|
|
|
|
if (cpu_dai->resume && !cpu_dai->ac97_control)
|
|
cpu_dai->resume(cpu_dai);
|
|
if (platform->resume)
|
|
platform->resume(&card->dai_link[i]);
|
|
}
|
|
|
|
if (card->resume_post)
|
|
card->resume_post(pdev);
|
|
|
|
dev_dbg(socdev->dev, "resume work completed\n");
|
|
|
|
/* userspace can access us now we are back as we were before */
|
|
snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0);
|
|
}
|
|
|
|
/* powers up audio subsystem after a suspend */
|
|
static int soc_resume(struct device *dev)
|
|
{
|
|
struct platform_device *pdev = to_platform_device(dev);
|
|
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
|
|
struct snd_soc_card *card = socdev->card;
|
|
struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai;
|
|
|
|
/* If the initialization of this soc device failed, there is no codec
|
|
* associated with it. Just bail out in this case.
|
|
*/
|
|
if (!card->codec)
|
|
return 0;
|
|
|
|
/* AC97 devices might have other drivers hanging off them so
|
|
* need to resume immediately. Other drivers don't have that
|
|
* problem and may take a substantial amount of time to resume
|
|
* due to I/O costs and anti-pop so handle them out of line.
|
|
*/
|
|
if (cpu_dai->ac97_control) {
|
|
dev_dbg(socdev->dev, "Resuming AC97 immediately\n");
|
|
soc_resume_deferred(&card->deferred_resume_work);
|
|
} else {
|
|
dev_dbg(socdev->dev, "Scheduling resume work\n");
|
|
if (!schedule_work(&card->deferred_resume_work))
|
|
dev_err(socdev->dev, "resume work item may be lost\n");
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
#else
|
|
#define soc_suspend NULL
|
|
#define soc_resume NULL
|
|
#endif
|
|
|
|
static struct snd_soc_dai_ops null_dai_ops = {
|
|
};
|
|
|
|
static void snd_soc_instantiate_card(struct snd_soc_card *card)
|
|
{
|
|
struct platform_device *pdev = container_of(card->dev,
|
|
struct platform_device,
|
|
dev);
|
|
struct snd_soc_codec_device *codec_dev = card->socdev->codec_dev;
|
|
struct snd_soc_codec *codec;
|
|
struct snd_soc_platform *platform;
|
|
struct snd_soc_dai *dai;
|
|
int i, found, ret, ac97;
|
|
|
|
if (card->instantiated)
|
|
return;
|
|
|
|
found = 0;
|
|
list_for_each_entry(platform, &platform_list, list)
|
|
if (card->platform == platform) {
|
|
found = 1;
|
|
break;
|
|
}
|
|
if (!found) {
|
|
dev_dbg(card->dev, "Platform %s not registered\n",
|
|
card->platform->name);
|
|
return;
|
|
}
|
|
|
|
ac97 = 0;
|
|
for (i = 0; i < card->num_links; i++) {
|
|
found = 0;
|
|
list_for_each_entry(dai, &dai_list, list)
|
|
if (card->dai_link[i].cpu_dai == dai) {
|
|
found = 1;
|
|
break;
|
|
}
|
|
if (!found) {
|
|
dev_dbg(card->dev, "DAI %s not registered\n",
|
|
card->dai_link[i].cpu_dai->name);
|
|
return;
|
|
}
|
|
|
|
if (card->dai_link[i].cpu_dai->ac97_control)
|
|
ac97 = 1;
|
|
}
|
|
|
|
for (i = 0; i < card->num_links; i++) {
|
|
if (!card->dai_link[i].codec_dai->ops)
|
|
card->dai_link[i].codec_dai->ops = &null_dai_ops;
|
|
}
|
|
|
|
/* If we have AC97 in the system then don't wait for the
|
|
* codec. This will need revisiting if we have to handle
|
|
* systems with mixed AC97 and non-AC97 parts. Only check for
|
|
* DAIs currently; we can't do this per link since some AC97
|
|
* codecs have non-AC97 DAIs.
|
|
*/
|
|
if (!ac97)
|
|
for (i = 0; i < card->num_links; i++) {
|
|
found = 0;
|
|
list_for_each_entry(dai, &dai_list, list)
|
|
if (card->dai_link[i].codec_dai == dai) {
|
|
found = 1;
|
|
break;
|
|
}
|
|
if (!found) {
|
|
dev_dbg(card->dev, "DAI %s not registered\n",
|
|
card->dai_link[i].codec_dai->name);
|
|
return;
|
|
}
|
|
}
|
|
|
|
/* Note that we do not current check for codec components */
|
|
|
|
dev_dbg(card->dev, "All components present, instantiating\n");
|
|
|
|
/* Found everything, bring it up */
|
|
card->pmdown_time = pmdown_time;
|
|
|
|
if (card->probe) {
|
|
ret = card->probe(pdev);
|
|
if (ret < 0)
|
|
return;
|
|
}
|
|
|
|
for (i = 0; i < card->num_links; i++) {
|
|
struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
|
|
if (cpu_dai->probe) {
|
|
ret = cpu_dai->probe(pdev, cpu_dai);
|
|
if (ret < 0)
|
|
goto cpu_dai_err;
|
|
}
|
|
}
|
|
|
|
if (codec_dev->probe) {
|
|
ret = codec_dev->probe(pdev);
|
|
if (ret < 0)
|
|
goto cpu_dai_err;
|
|
}
|
|
codec = card->codec;
|
|
|
|
if (platform->probe) {
|
|
ret = platform->probe(pdev);
|
|
if (ret < 0)
|
|
goto platform_err;
|
|
}
|
|
|
|
/* DAPM stream work */
|
|
INIT_DELAYED_WORK(&card->delayed_work, close_delayed_work);
|
|
#ifdef CONFIG_PM
|
|
/* deferred resume work */
|
|
INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
|
|
#endif
|
|
|
|
for (i = 0; i < card->num_links; i++) {
|
|
if (card->dai_link[i].init) {
|
|
ret = card->dai_link[i].init(codec);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: failed to init %s\n",
|
|
card->dai_link[i].stream_name);
|
|
continue;
|
|
}
|
|
}
|
|
if (card->dai_link[i].codec_dai->ac97_control)
|
|
ac97 = 1;
|
|
}
|
|
|
|
snprintf(codec->card->shortname, sizeof(codec->card->shortname),
|
|
"%s", card->name);
|
|
snprintf(codec->card->longname, sizeof(codec->card->longname),
|
|
"%s (%s)", card->name, codec->name);
|
|
|
|
/* Make sure all DAPM widgets are instantiated */
|
|
snd_soc_dapm_new_widgets(codec);
|
|
|
|
ret = snd_card_register(codec->card);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: failed to register soundcard for %s\n",
|
|
codec->name);
|
|
goto card_err;
|
|
}
|
|
|
|
mutex_lock(&codec->mutex);
|
|
#ifdef CONFIG_SND_SOC_AC97_BUS
|
|
/* Only instantiate AC97 if not already done by the adaptor
|
|
* for the generic AC97 subsystem.
|
|
*/
|
|
if (ac97 && strcmp(codec->name, "AC97") != 0) {
|
|
ret = soc_ac97_dev_register(codec);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: AC97 device register failed\n");
|
|
snd_card_free(codec->card);
|
|
mutex_unlock(&codec->mutex);
|
|
goto card_err;
|
|
}
|
|
}
|
|
#endif
|
|
|
|
ret = snd_soc_dapm_sys_add(card->socdev->dev);
|
|
if (ret < 0)
|
|
printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n");
|
|
|
|
ret = device_create_file(card->socdev->dev, &dev_attr_pmdown_time);
|
|
if (ret < 0)
|
|
printk(KERN_WARNING "asoc: failed to add pmdown_time sysfs\n");
|
|
|
|
ret = device_create_file(card->socdev->dev, &dev_attr_codec_reg);
|
|
if (ret < 0)
|
|
printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
|
|
|
|
soc_init_codec_debugfs(codec);
|
|
mutex_unlock(&codec->mutex);
|
|
|
|
card->instantiated = 1;
|
|
|
|
return;
|
|
|
|
card_err:
|
|
if (platform->remove)
|
|
platform->remove(pdev);
|
|
|
|
platform_err:
|
|
if (codec_dev->remove)
|
|
codec_dev->remove(pdev);
|
|
|
|
cpu_dai_err:
|
|
for (i--; i >= 0; i--) {
|
|
struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
|
|
if (cpu_dai->remove)
|
|
cpu_dai->remove(pdev, cpu_dai);
|
|
}
|
|
|
|
if (card->remove)
|
|
card->remove(pdev);
|
|
}
|
|
|
|
/*
|
|
* Attempt to initialise any uninitialised cards. Must be called with
|
|
* client_mutex.
|
|
*/
|
|
static void snd_soc_instantiate_cards(void)
|
|
{
|
|
struct snd_soc_card *card;
|
|
list_for_each_entry(card, &card_list, list)
|
|
snd_soc_instantiate_card(card);
|
|
}
|
|
|
|
/* probes a new socdev */
|
|
static int soc_probe(struct platform_device *pdev)
|
|
{
|
|
int ret = 0;
|
|
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
|
|
struct snd_soc_card *card = socdev->card;
|
|
|
|
/* Bodge while we push things out of socdev */
|
|
card->socdev = socdev;
|
|
|
|
/* Bodge while we unpick instantiation */
|
|
card->dev = &pdev->dev;
|
|
ret = snd_soc_register_card(card);
|
|
if (ret != 0) {
|
|
dev_err(&pdev->dev, "Failed to register card\n");
|
|
return ret;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* removes a socdev */
|
|
static int soc_remove(struct platform_device *pdev)
|
|
{
|
|
int i;
|
|
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
|
|
struct snd_soc_card *card = socdev->card;
|
|
struct snd_soc_platform *platform = card->platform;
|
|
struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
|
|
|
|
if (card->instantiated) {
|
|
run_delayed_work(&card->delayed_work);
|
|
|
|
if (platform->remove)
|
|
platform->remove(pdev);
|
|
|
|
if (codec_dev->remove)
|
|
codec_dev->remove(pdev);
|
|
|
|
for (i = 0; i < card->num_links; i++) {
|
|
struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
|
|
if (cpu_dai->remove)
|
|
cpu_dai->remove(pdev, cpu_dai);
|
|
}
|
|
|
|
if (card->remove)
|
|
card->remove(pdev);
|
|
}
|
|
|
|
snd_soc_unregister_card(card);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int soc_poweroff(struct device *dev)
|
|
{
|
|
struct platform_device *pdev = to_platform_device(dev);
|
|
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
|
|
struct snd_soc_card *card = socdev->card;
|
|
|
|
if (!card->instantiated)
|
|
return 0;
|
|
|
|
/* Flush out pmdown_time work - we actually do want to run it
|
|
* now, we're shutting down so no imminent restart. */
|
|
run_delayed_work(&card->delayed_work);
|
|
|
|
snd_soc_dapm_shutdown(socdev);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static const struct dev_pm_ops soc_pm_ops = {
|
|
.suspend = soc_suspend,
|
|
.resume = soc_resume,
|
|
.poweroff = soc_poweroff,
|
|
};
|
|
|
|
/* ASoC platform driver */
|
|
static struct platform_driver soc_driver = {
|
|
.driver = {
|
|
.name = "soc-audio",
|
|
.owner = THIS_MODULE,
|
|
.pm = &soc_pm_ops,
|
|
},
|
|
.probe = soc_probe,
|
|
.remove = soc_remove,
|
|
};
|
|
|
|
/* create a new pcm */
|
|
static int soc_new_pcm(struct snd_soc_device *socdev,
|
|
struct snd_soc_dai_link *dai_link, int num)
|
|
{
|
|
struct snd_soc_card *card = socdev->card;
|
|
struct snd_soc_codec *codec = card->codec;
|
|
struct snd_soc_platform *platform = card->platform;
|
|
struct snd_soc_dai *codec_dai = dai_link->codec_dai;
|
|
struct snd_soc_dai *cpu_dai = dai_link->cpu_dai;
|
|
struct snd_soc_pcm_runtime *rtd;
|
|
struct snd_pcm *pcm;
|
|
char new_name[64];
|
|
int ret = 0, playback = 0, capture = 0;
|
|
|
|
rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL);
|
|
if (rtd == NULL)
|
|
return -ENOMEM;
|
|
|
|
rtd->dai = dai_link;
|
|
rtd->socdev = socdev;
|
|
codec_dai->codec = card->codec;
|
|
|
|
/* check client and interface hw capabilities */
|
|
snprintf(new_name, sizeof(new_name), "%s %s-%d",
|
|
dai_link->stream_name, codec_dai->name, num);
|
|
|
|
if (codec_dai->playback.channels_min)
|
|
playback = 1;
|
|
if (codec_dai->capture.channels_min)
|
|
capture = 1;
|
|
|
|
ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback,
|
|
capture, &pcm);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: can't create pcm for codec %s\n",
|
|
codec->name);
|
|
kfree(rtd);
|
|
return ret;
|
|
}
|
|
|
|
dai_link->pcm = pcm;
|
|
pcm->private_data = rtd;
|
|
soc_pcm_ops.mmap = platform->pcm_ops->mmap;
|
|
soc_pcm_ops.ioctl = platform->pcm_ops->ioctl;
|
|
soc_pcm_ops.copy = platform->pcm_ops->copy;
|
|
soc_pcm_ops.silence = platform->pcm_ops->silence;
|
|
soc_pcm_ops.ack = platform->pcm_ops->ack;
|
|
soc_pcm_ops.page = platform->pcm_ops->page;
|
|
|
|
if (playback)
|
|
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops);
|
|
|
|
if (capture)
|
|
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops);
|
|
|
|
ret = platform->pcm_new(codec->card, codec_dai, pcm);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: platform pcm constructor failed\n");
|
|
kfree(rtd);
|
|
return ret;
|
|
}
|
|
|
|
pcm->private_free = platform->pcm_free;
|
|
printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
|
|
cpu_dai->name);
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* snd_soc_codec_volatile_register: Report if a register is volatile.
|
|
*
|
|
* @codec: CODEC to query.
|
|
* @reg: Register to query.
|
|
*
|
|
* Boolean function indiciating if a CODEC register is volatile.
|
|
*/
|
|
int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg)
|
|
{
|
|
if (codec->volatile_register)
|
|
return codec->volatile_register(reg);
|
|
else
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_codec_volatile_register);
|
|
|
|
/**
|
|
* snd_soc_new_ac97_codec - initailise AC97 device
|
|
* @codec: audio codec
|
|
* @ops: AC97 bus operations
|
|
* @num: AC97 codec number
|
|
*
|
|
* Initialises AC97 codec resources for use by ad-hoc devices only.
|
|
*/
|
|
int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
|
|
struct snd_ac97_bus_ops *ops, int num)
|
|
{
|
|
mutex_lock(&codec->mutex);
|
|
|
|
codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL);
|
|
if (codec->ac97 == NULL) {
|
|
mutex_unlock(&codec->mutex);
|
|
return -ENOMEM;
|
|
}
|
|
|
|
codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL);
|
|
if (codec->ac97->bus == NULL) {
|
|
kfree(codec->ac97);
|
|
codec->ac97 = NULL;
|
|
mutex_unlock(&codec->mutex);
|
|
return -ENOMEM;
|
|
}
|
|
|
|
codec->ac97->bus->ops = ops;
|
|
codec->ac97->num = num;
|
|
codec->dev = &codec->ac97->dev;
|
|
mutex_unlock(&codec->mutex);
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec);
|
|
|
|
/**
|
|
* snd_soc_free_ac97_codec - free AC97 codec device
|
|
* @codec: audio codec
|
|
*
|
|
* Frees AC97 codec device resources.
|
|
*/
|
|
void snd_soc_free_ac97_codec(struct snd_soc_codec *codec)
|
|
{
|
|
mutex_lock(&codec->mutex);
|
|
kfree(codec->ac97->bus);
|
|
kfree(codec->ac97);
|
|
codec->ac97 = NULL;
|
|
mutex_unlock(&codec->mutex);
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec);
|
|
|
|
/**
|
|
* snd_soc_update_bits - update codec register bits
|
|
* @codec: audio codec
|
|
* @reg: codec register
|
|
* @mask: register mask
|
|
* @value: new value
|
|
*
|
|
* Writes new register value.
|
|
*
|
|
* Returns 1 for change else 0.
|
|
*/
|
|
int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
|
|
unsigned int mask, unsigned int value)
|
|
{
|
|
int change;
|
|
unsigned int old, new;
|
|
|
|
old = snd_soc_read(codec, reg);
|
|
new = (old & ~mask) | value;
|
|
change = old != new;
|
|
if (change)
|
|
snd_soc_write(codec, reg, new);
|
|
|
|
return change;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_update_bits);
|
|
|
|
/**
|
|
* snd_soc_update_bits_locked - update codec register bits
|
|
* @codec: audio codec
|
|
* @reg: codec register
|
|
* @mask: register mask
|
|
* @value: new value
|
|
*
|
|
* Writes new register value, and takes the codec mutex.
|
|
*
|
|
* Returns 1 for change else 0.
|
|
*/
|
|
int snd_soc_update_bits_locked(struct snd_soc_codec *codec,
|
|
unsigned short reg, unsigned int mask,
|
|
unsigned int value)
|
|
{
|
|
int change;
|
|
|
|
mutex_lock(&codec->mutex);
|
|
change = snd_soc_update_bits(codec, reg, mask, value);
|
|
mutex_unlock(&codec->mutex);
|
|
|
|
return change;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_update_bits_locked);
|
|
|
|
/**
|
|
* snd_soc_test_bits - test register for change
|
|
* @codec: audio codec
|
|
* @reg: codec register
|
|
* @mask: register mask
|
|
* @value: new value
|
|
*
|
|
* Tests a register with a new value and checks if the new value is
|
|
* different from the old value.
|
|
*
|
|
* Returns 1 for change else 0.
|
|
*/
|
|
int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg,
|
|
unsigned int mask, unsigned int value)
|
|
{
|
|
int change;
|
|
unsigned int old, new;
|
|
|
|
old = snd_soc_read(codec, reg);
|
|
new = (old & ~mask) | value;
|
|
change = old != new;
|
|
|
|
return change;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_test_bits);
|
|
|
|
/**
|
|
* snd_soc_new_pcms - create new sound card and pcms
|
|
* @socdev: the SoC audio device
|
|
* @idx: ALSA card index
|
|
* @xid: card identification
|
|
*
|
|
* Create a new sound card based upon the codec and interface pcms.
|
|
*
|
|
* Returns 0 for success, else error.
|
|
*/
|
|
int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
|
|
{
|
|
struct snd_soc_card *card = socdev->card;
|
|
struct snd_soc_codec *codec = card->codec;
|
|
int ret, i;
|
|
|
|
mutex_lock(&codec->mutex);
|
|
|
|
/* register a sound card */
|
|
ret = snd_card_create(idx, xid, codec->owner, 0, &codec->card);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: can't create sound card for codec %s\n",
|
|
codec->name);
|
|
mutex_unlock(&codec->mutex);
|
|
return ret;
|
|
}
|
|
|
|
codec->socdev = socdev;
|
|
codec->card->dev = socdev->dev;
|
|
codec->card->private_data = codec;
|
|
strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver));
|
|
|
|
/* create the pcms */
|
|
for (i = 0; i < card->num_links; i++) {
|
|
ret = soc_new_pcm(socdev, &card->dai_link[i], i);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "asoc: can't create pcm %s\n",
|
|
card->dai_link[i].stream_name);
|
|
mutex_unlock(&codec->mutex);
|
|
return ret;
|
|
}
|
|
/* Check for codec->ac97 to handle the ac97.c fun */
|
|
if (card->dai_link[i].codec_dai->ac97_control && codec->ac97) {
|
|
snd_ac97_dev_add_pdata(codec->ac97,
|
|
card->dai_link[i].cpu_dai->ac97_pdata);
|
|
}
|
|
}
|
|
|
|
mutex_unlock(&codec->mutex);
|
|
return ret;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_new_pcms);
|
|
|
|
/**
|
|
* snd_soc_free_pcms - free sound card and pcms
|
|
* @socdev: the SoC audio device
|
|
*
|
|
* Frees sound card and pcms associated with the socdev.
|
|
* Also unregister the codec if it is an AC97 device.
|
|
*/
|
|
void snd_soc_free_pcms(struct snd_soc_device *socdev)
|
|
{
|
|
struct snd_soc_codec *codec = socdev->card->codec;
|
|
#ifdef CONFIG_SND_SOC_AC97_BUS
|
|
struct snd_soc_dai *codec_dai;
|
|
int i;
|
|
#endif
|
|
|
|
mutex_lock(&codec->mutex);
|
|
soc_cleanup_codec_debugfs(codec);
|
|
#ifdef CONFIG_SND_SOC_AC97_BUS
|
|
for (i = 0; i < codec->num_dai; i++) {
|
|
codec_dai = &codec->dai[i];
|
|
if (codec_dai->ac97_control && codec->ac97 &&
|
|
strcmp(codec->name, "AC97") != 0) {
|
|
soc_ac97_dev_unregister(codec);
|
|
goto free_card;
|
|
}
|
|
}
|
|
free_card:
|
|
#endif
|
|
|
|
if (codec->card)
|
|
snd_card_free(codec->card);
|
|
device_remove_file(socdev->dev, &dev_attr_codec_reg);
|
|
mutex_unlock(&codec->mutex);
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_free_pcms);
|
|
|
|
/**
|
|
* snd_soc_set_runtime_hwparams - set the runtime hardware parameters
|
|
* @substream: the pcm substream
|
|
* @hw: the hardware parameters
|
|
*
|
|
* Sets the substream runtime hardware parameters.
|
|
*/
|
|
int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
|
|
const struct snd_pcm_hardware *hw)
|
|
{
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
runtime->hw.info = hw->info;
|
|
runtime->hw.formats = hw->formats;
|
|
runtime->hw.period_bytes_min = hw->period_bytes_min;
|
|
runtime->hw.period_bytes_max = hw->period_bytes_max;
|
|
runtime->hw.periods_min = hw->periods_min;
|
|
runtime->hw.periods_max = hw->periods_max;
|
|
runtime->hw.buffer_bytes_max = hw->buffer_bytes_max;
|
|
runtime->hw.fifo_size = hw->fifo_size;
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams);
|
|
|
|
/**
|
|
* snd_soc_cnew - create new control
|
|
* @_template: control template
|
|
* @data: control private data
|
|
* @long_name: control long name
|
|
*
|
|
* Create a new mixer control from a template control.
|
|
*
|
|
* Returns 0 for success, else error.
|
|
*/
|
|
struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
|
|
void *data, char *long_name)
|
|
{
|
|
struct snd_kcontrol_new template;
|
|
|
|
memcpy(&template, _template, sizeof(template));
|
|
if (long_name)
|
|
template.name = long_name;
|
|
template.index = 0;
|
|
|
|
return snd_ctl_new1(&template, data);
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_cnew);
|
|
|
|
/**
|
|
* snd_soc_add_controls - add an array of controls to a codec.
|
|
* Convienience function to add a list of controls. Many codecs were
|
|
* duplicating this code.
|
|
*
|
|
* @codec: codec to add controls to
|
|
* @controls: array of controls to add
|
|
* @num_controls: number of elements in the array
|
|
*
|
|
* Return 0 for success, else error.
|
|
*/
|
|
int snd_soc_add_controls(struct snd_soc_codec *codec,
|
|
const struct snd_kcontrol_new *controls, int num_controls)
|
|
{
|
|
struct snd_card *card = codec->card;
|
|
int err, i;
|
|
|
|
for (i = 0; i < num_controls; i++) {
|
|
const struct snd_kcontrol_new *control = &controls[i];
|
|
err = snd_ctl_add(card, snd_soc_cnew(control, codec, NULL));
|
|
if (err < 0) {
|
|
dev_err(codec->dev, "%s: Failed to add %s\n",
|
|
codec->name, control->name);
|
|
return err;
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_add_controls);
|
|
|
|
/**
|
|
* snd_soc_info_enum_double - enumerated double mixer info callback
|
|
* @kcontrol: mixer control
|
|
* @uinfo: control element information
|
|
*
|
|
* Callback to provide information about a double enumerated
|
|
* mixer control.
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_info *uinfo)
|
|
{
|
|
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
|
|
|
|
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
|
|
uinfo->count = e->shift_l == e->shift_r ? 1 : 2;
|
|
uinfo->value.enumerated.items = e->max;
|
|
|
|
if (uinfo->value.enumerated.item > e->max - 1)
|
|
uinfo->value.enumerated.item = e->max - 1;
|
|
strcpy(uinfo->value.enumerated.name,
|
|
e->texts[uinfo->value.enumerated.item]);
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_info_enum_double);
|
|
|
|
/**
|
|
* snd_soc_get_enum_double - enumerated double mixer get callback
|
|
* @kcontrol: mixer control
|
|
* @ucontrol: control element information
|
|
*
|
|
* Callback to get the value of a double enumerated mixer.
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_value *ucontrol)
|
|
{
|
|
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
|
|
unsigned int val, bitmask;
|
|
|
|
for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
|
|
;
|
|
val = snd_soc_read(codec, e->reg);
|
|
ucontrol->value.enumerated.item[0]
|
|
= (val >> e->shift_l) & (bitmask - 1);
|
|
if (e->shift_l != e->shift_r)
|
|
ucontrol->value.enumerated.item[1] =
|
|
(val >> e->shift_r) & (bitmask - 1);
|
|
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_get_enum_double);
|
|
|
|
/**
|
|
* snd_soc_put_enum_double - enumerated double mixer put callback
|
|
* @kcontrol: mixer control
|
|
* @ucontrol: control element information
|
|
*
|
|
* Callback to set the value of a double enumerated mixer.
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_value *ucontrol)
|
|
{
|
|
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
|
|
unsigned int val;
|
|
unsigned int mask, bitmask;
|
|
|
|
for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
|
|
;
|
|
if (ucontrol->value.enumerated.item[0] > e->max - 1)
|
|
return -EINVAL;
|
|
val = ucontrol->value.enumerated.item[0] << e->shift_l;
|
|
mask = (bitmask - 1) << e->shift_l;
|
|
if (e->shift_l != e->shift_r) {
|
|
if (ucontrol->value.enumerated.item[1] > e->max - 1)
|
|
return -EINVAL;
|
|
val |= ucontrol->value.enumerated.item[1] << e->shift_r;
|
|
mask |= (bitmask - 1) << e->shift_r;
|
|
}
|
|
|
|
return snd_soc_update_bits_locked(codec, e->reg, mask, val);
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_put_enum_double);
|
|
|
|
/**
|
|
* snd_soc_get_value_enum_double - semi enumerated double mixer get callback
|
|
* @kcontrol: mixer control
|
|
* @ucontrol: control element information
|
|
*
|
|
* Callback to get the value of a double semi enumerated mixer.
|
|
*
|
|
* Semi enumerated mixer: the enumerated items are referred as values. Can be
|
|
* used for handling bitfield coded enumeration for example.
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_value *ucontrol)
|
|
{
|
|
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
|
|
unsigned int reg_val, val, mux;
|
|
|
|
reg_val = snd_soc_read(codec, e->reg);
|
|
val = (reg_val >> e->shift_l) & e->mask;
|
|
for (mux = 0; mux < e->max; mux++) {
|
|
if (val == e->values[mux])
|
|
break;
|
|
}
|
|
ucontrol->value.enumerated.item[0] = mux;
|
|
if (e->shift_l != e->shift_r) {
|
|
val = (reg_val >> e->shift_r) & e->mask;
|
|
for (mux = 0; mux < e->max; mux++) {
|
|
if (val == e->values[mux])
|
|
break;
|
|
}
|
|
ucontrol->value.enumerated.item[1] = mux;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_get_value_enum_double);
|
|
|
|
/**
|
|
* snd_soc_put_value_enum_double - semi enumerated double mixer put callback
|
|
* @kcontrol: mixer control
|
|
* @ucontrol: control element information
|
|
*
|
|
* Callback to set the value of a double semi enumerated mixer.
|
|
*
|
|
* Semi enumerated mixer: the enumerated items are referred as values. Can be
|
|
* used for handling bitfield coded enumeration for example.
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_value *ucontrol)
|
|
{
|
|
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
|
|
unsigned int val;
|
|
unsigned int mask;
|
|
|
|
if (ucontrol->value.enumerated.item[0] > e->max - 1)
|
|
return -EINVAL;
|
|
val = e->values[ucontrol->value.enumerated.item[0]] << e->shift_l;
|
|
mask = e->mask << e->shift_l;
|
|
if (e->shift_l != e->shift_r) {
|
|
if (ucontrol->value.enumerated.item[1] > e->max - 1)
|
|
return -EINVAL;
|
|
val |= e->values[ucontrol->value.enumerated.item[1]] << e->shift_r;
|
|
mask |= e->mask << e->shift_r;
|
|
}
|
|
|
|
return snd_soc_update_bits_locked(codec, e->reg, mask, val);
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_put_value_enum_double);
|
|
|
|
/**
|
|
* snd_soc_info_enum_ext - external enumerated single mixer info callback
|
|
* @kcontrol: mixer control
|
|
* @uinfo: control element information
|
|
*
|
|
* Callback to provide information about an external enumerated
|
|
* single mixer.
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_info *uinfo)
|
|
{
|
|
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
|
|
|
|
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
|
|
uinfo->count = 1;
|
|
uinfo->value.enumerated.items = e->max;
|
|
|
|
if (uinfo->value.enumerated.item > e->max - 1)
|
|
uinfo->value.enumerated.item = e->max - 1;
|
|
strcpy(uinfo->value.enumerated.name,
|
|
e->texts[uinfo->value.enumerated.item]);
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext);
|
|
|
|
/**
|
|
* snd_soc_info_volsw_ext - external single mixer info callback
|
|
* @kcontrol: mixer control
|
|
* @uinfo: control element information
|
|
*
|
|
* Callback to provide information about a single external mixer control.
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_info *uinfo)
|
|
{
|
|
int max = kcontrol->private_value;
|
|
|
|
if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
|
|
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
|
|
else
|
|
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
|
|
|
|
uinfo->count = 1;
|
|
uinfo->value.integer.min = 0;
|
|
uinfo->value.integer.max = max;
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext);
|
|
|
|
/**
|
|
* snd_soc_info_volsw - single mixer info callback
|
|
* @kcontrol: mixer control
|
|
* @uinfo: control element information
|
|
*
|
|
* Callback to provide information about a single mixer control.
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_info *uinfo)
|
|
{
|
|
struct soc_mixer_control *mc =
|
|
(struct soc_mixer_control *)kcontrol->private_value;
|
|
int platform_max;
|
|
unsigned int shift = mc->shift;
|
|
unsigned int rshift = mc->rshift;
|
|
|
|
if (!mc->platform_max)
|
|
mc->platform_max = mc->max;
|
|
platform_max = mc->platform_max;
|
|
|
|
if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume"))
|
|
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
|
|
else
|
|
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
|
|
|
|
uinfo->count = shift == rshift ? 1 : 2;
|
|
uinfo->value.integer.min = 0;
|
|
uinfo->value.integer.max = platform_max;
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_info_volsw);
|
|
|
|
/**
|
|
* snd_soc_get_volsw - single mixer get callback
|
|
* @kcontrol: mixer control
|
|
* @ucontrol: control element information
|
|
*
|
|
* Callback to get the value of a single mixer control.
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_value *ucontrol)
|
|
{
|
|
struct soc_mixer_control *mc =
|
|
(struct soc_mixer_control *)kcontrol->private_value;
|
|
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
unsigned int reg = mc->reg;
|
|
unsigned int shift = mc->shift;
|
|
unsigned int rshift = mc->rshift;
|
|
int max = mc->max;
|
|
unsigned int mask = (1 << fls(max)) - 1;
|
|
unsigned int invert = mc->invert;
|
|
|
|
ucontrol->value.integer.value[0] =
|
|
(snd_soc_read(codec, reg) >> shift) & mask;
|
|
if (shift != rshift)
|
|
ucontrol->value.integer.value[1] =
|
|
(snd_soc_read(codec, reg) >> rshift) & mask;
|
|
if (invert) {
|
|
ucontrol->value.integer.value[0] =
|
|
max - ucontrol->value.integer.value[0];
|
|
if (shift != rshift)
|
|
ucontrol->value.integer.value[1] =
|
|
max - ucontrol->value.integer.value[1];
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_get_volsw);
|
|
|
|
/**
|
|
* snd_soc_put_volsw - single mixer put callback
|
|
* @kcontrol: mixer control
|
|
* @ucontrol: control element information
|
|
*
|
|
* Callback to set the value of a single mixer control.
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_value *ucontrol)
|
|
{
|
|
struct soc_mixer_control *mc =
|
|
(struct soc_mixer_control *)kcontrol->private_value;
|
|
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
unsigned int reg = mc->reg;
|
|
unsigned int shift = mc->shift;
|
|
unsigned int rshift = mc->rshift;
|
|
int max = mc->max;
|
|
unsigned int mask = (1 << fls(max)) - 1;
|
|
unsigned int invert = mc->invert;
|
|
unsigned int val, val2, val_mask;
|
|
|
|
val = (ucontrol->value.integer.value[0] & mask);
|
|
if (invert)
|
|
val = max - val;
|
|
val_mask = mask << shift;
|
|
val = val << shift;
|
|
if (shift != rshift) {
|
|
val2 = (ucontrol->value.integer.value[1] & mask);
|
|
if (invert)
|
|
val2 = max - val2;
|
|
val_mask |= mask << rshift;
|
|
val |= val2 << rshift;
|
|
}
|
|
return snd_soc_update_bits_locked(codec, reg, val_mask, val);
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_put_volsw);
|
|
|
|
/**
|
|
* snd_soc_info_volsw_2r - double mixer info callback
|
|
* @kcontrol: mixer control
|
|
* @uinfo: control element information
|
|
*
|
|
* Callback to provide information about a double mixer control that
|
|
* spans 2 codec registers.
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_info *uinfo)
|
|
{
|
|
struct soc_mixer_control *mc =
|
|
(struct soc_mixer_control *)kcontrol->private_value;
|
|
int platform_max;
|
|
|
|
if (!mc->platform_max)
|
|
mc->platform_max = mc->max;
|
|
platform_max = mc->platform_max;
|
|
|
|
if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume"))
|
|
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
|
|
else
|
|
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
|
|
|
|
uinfo->count = 2;
|
|
uinfo->value.integer.min = 0;
|
|
uinfo->value.integer.max = platform_max;
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r);
|
|
|
|
/**
|
|
* snd_soc_get_volsw_2r - double mixer get callback
|
|
* @kcontrol: mixer control
|
|
* @ucontrol: control element information
|
|
*
|
|
* Callback to get the value of a double mixer control that spans 2 registers.
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_value *ucontrol)
|
|
{
|
|
struct soc_mixer_control *mc =
|
|
(struct soc_mixer_control *)kcontrol->private_value;
|
|
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
unsigned int reg = mc->reg;
|
|
unsigned int reg2 = mc->rreg;
|
|
unsigned int shift = mc->shift;
|
|
int max = mc->max;
|
|
unsigned int mask = (1 << fls(max)) - 1;
|
|
unsigned int invert = mc->invert;
|
|
|
|
ucontrol->value.integer.value[0] =
|
|
(snd_soc_read(codec, reg) >> shift) & mask;
|
|
ucontrol->value.integer.value[1] =
|
|
(snd_soc_read(codec, reg2) >> shift) & mask;
|
|
if (invert) {
|
|
ucontrol->value.integer.value[0] =
|
|
max - ucontrol->value.integer.value[0];
|
|
ucontrol->value.integer.value[1] =
|
|
max - ucontrol->value.integer.value[1];
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r);
|
|
|
|
/**
|
|
* snd_soc_put_volsw_2r - double mixer set callback
|
|
* @kcontrol: mixer control
|
|
* @ucontrol: control element information
|
|
*
|
|
* Callback to set the value of a double mixer control that spans 2 registers.
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_value *ucontrol)
|
|
{
|
|
struct soc_mixer_control *mc =
|
|
(struct soc_mixer_control *)kcontrol->private_value;
|
|
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
unsigned int reg = mc->reg;
|
|
unsigned int reg2 = mc->rreg;
|
|
unsigned int shift = mc->shift;
|
|
int max = mc->max;
|
|
unsigned int mask = (1 << fls(max)) - 1;
|
|
unsigned int invert = mc->invert;
|
|
int err;
|
|
unsigned int val, val2, val_mask;
|
|
|
|
val_mask = mask << shift;
|
|
val = (ucontrol->value.integer.value[0] & mask);
|
|
val2 = (ucontrol->value.integer.value[1] & mask);
|
|
|
|
if (invert) {
|
|
val = max - val;
|
|
val2 = max - val2;
|
|
}
|
|
|
|
val = val << shift;
|
|
val2 = val2 << shift;
|
|
|
|
err = snd_soc_update_bits_locked(codec, reg, val_mask, val);
|
|
if (err < 0)
|
|
return err;
|
|
|
|
err = snd_soc_update_bits_locked(codec, reg2, val_mask, val2);
|
|
return err;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r);
|
|
|
|
/**
|
|
* snd_soc_info_volsw_s8 - signed mixer info callback
|
|
* @kcontrol: mixer control
|
|
* @uinfo: control element information
|
|
*
|
|
* Callback to provide information about a signed mixer control.
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_info *uinfo)
|
|
{
|
|
struct soc_mixer_control *mc =
|
|
(struct soc_mixer_control *)kcontrol->private_value;
|
|
int platform_max;
|
|
int min = mc->min;
|
|
|
|
if (!mc->platform_max)
|
|
mc->platform_max = mc->max;
|
|
platform_max = mc->platform_max;
|
|
|
|
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
|
|
uinfo->count = 2;
|
|
uinfo->value.integer.min = 0;
|
|
uinfo->value.integer.max = platform_max - min;
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8);
|
|
|
|
/**
|
|
* snd_soc_get_volsw_s8 - signed mixer get callback
|
|
* @kcontrol: mixer control
|
|
* @ucontrol: control element information
|
|
*
|
|
* Callback to get the value of a signed mixer control.
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_value *ucontrol)
|
|
{
|
|
struct soc_mixer_control *mc =
|
|
(struct soc_mixer_control *)kcontrol->private_value;
|
|
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
unsigned int reg = mc->reg;
|
|
int min = mc->min;
|
|
int val = snd_soc_read(codec, reg);
|
|
|
|
ucontrol->value.integer.value[0] =
|
|
((signed char)(val & 0xff))-min;
|
|
ucontrol->value.integer.value[1] =
|
|
((signed char)((val >> 8) & 0xff))-min;
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8);
|
|
|
|
/**
|
|
* snd_soc_put_volsw_sgn - signed mixer put callback
|
|
* @kcontrol: mixer control
|
|
* @ucontrol: control element information
|
|
*
|
|
* Callback to set the value of a signed mixer control.
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_value *ucontrol)
|
|
{
|
|
struct soc_mixer_control *mc =
|
|
(struct soc_mixer_control *)kcontrol->private_value;
|
|
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
unsigned int reg = mc->reg;
|
|
int min = mc->min;
|
|
unsigned int val;
|
|
|
|
val = (ucontrol->value.integer.value[0]+min) & 0xff;
|
|
val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8;
|
|
|
|
return snd_soc_update_bits_locked(codec, reg, 0xffff, val);
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
|
|
|
|
/**
|
|
* snd_soc_limit_volume - Set new limit to an existing volume control.
|
|
*
|
|
* @codec: where to look for the control
|
|
* @name: Name of the control
|
|
* @max: new maximum limit
|
|
*
|
|
* Return 0 for success, else error.
|
|
*/
|
|
int snd_soc_limit_volume(struct snd_soc_codec *codec,
|
|
const char *name, int max)
|
|
{
|
|
struct snd_card *card = codec->card;
|
|
struct snd_kcontrol *kctl;
|
|
struct soc_mixer_control *mc;
|
|
int found = 0;
|
|
int ret = -EINVAL;
|
|
|
|
/* Sanity check for name and max */
|
|
if (unlikely(!name || max <= 0))
|
|
return -EINVAL;
|
|
|
|
list_for_each_entry(kctl, &card->controls, list) {
|
|
if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) {
|
|
found = 1;
|
|
break;
|
|
}
|
|
}
|
|
if (found) {
|
|
mc = (struct soc_mixer_control *)kctl->private_value;
|
|
if (max <= mc->max) {
|
|
mc->platform_max = max;
|
|
ret = 0;
|
|
}
|
|
}
|
|
return ret;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_limit_volume);
|
|
|
|
/**
|
|
* snd_soc_info_volsw_2r_sx - double with tlv and variable data size
|
|
* mixer info callback
|
|
* @kcontrol: mixer control
|
|
* @uinfo: control element information
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_info_volsw_2r_sx(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_info *uinfo)
|
|
{
|
|
struct soc_mixer_control *mc =
|
|
(struct soc_mixer_control *)kcontrol->private_value;
|
|
int max = mc->max;
|
|
int min = mc->min;
|
|
|
|
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
|
|
uinfo->count = 2;
|
|
uinfo->value.integer.min = 0;
|
|
uinfo->value.integer.max = max-min;
|
|
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r_sx);
|
|
|
|
/**
|
|
* snd_soc_get_volsw_2r_sx - double with tlv and variable data size
|
|
* mixer get callback
|
|
* @kcontrol: mixer control
|
|
* @uinfo: control element information
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_value *ucontrol)
|
|
{
|
|
struct soc_mixer_control *mc =
|
|
(struct soc_mixer_control *)kcontrol->private_value;
|
|
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
unsigned int mask = (1<<mc->shift)-1;
|
|
int min = mc->min;
|
|
int val = snd_soc_read(codec, mc->reg) & mask;
|
|
int valr = snd_soc_read(codec, mc->rreg) & mask;
|
|
|
|
ucontrol->value.integer.value[0] = ((val & 0xff)-min) & mask;
|
|
ucontrol->value.integer.value[1] = ((valr & 0xff)-min) & mask;
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r_sx);
|
|
|
|
/**
|
|
* snd_soc_put_volsw_2r_sx - double with tlv and variable data size
|
|
* mixer put callback
|
|
* @kcontrol: mixer control
|
|
* @uinfo: control element information
|
|
*
|
|
* Returns 0 for success.
|
|
*/
|
|
int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_value *ucontrol)
|
|
{
|
|
struct soc_mixer_control *mc =
|
|
(struct soc_mixer_control *)kcontrol->private_value;
|
|
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
|
|
unsigned int mask = (1<<mc->shift)-1;
|
|
int min = mc->min;
|
|
int ret;
|
|
unsigned int val, valr, oval, ovalr;
|
|
|
|
val = ((ucontrol->value.integer.value[0]+min) & 0xff);
|
|
val &= mask;
|
|
valr = ((ucontrol->value.integer.value[1]+min) & 0xff);
|
|
valr &= mask;
|
|
|
|
oval = snd_soc_read(codec, mc->reg) & mask;
|
|
ovalr = snd_soc_read(codec, mc->rreg) & mask;
|
|
|
|
ret = 0;
|
|
if (oval != val) {
|
|
ret = snd_soc_write(codec, mc->reg, val);
|
|
if (ret < 0)
|
|
return ret;
|
|
}
|
|
if (ovalr != valr) {
|
|
ret = snd_soc_write(codec, mc->rreg, valr);
|
|
if (ret < 0)
|
|
return ret;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r_sx);
|
|
|
|
/**
|
|
* snd_soc_dai_set_sysclk - configure DAI system or master clock.
|
|
* @dai: DAI
|
|
* @clk_id: DAI specific clock ID
|
|
* @freq: new clock frequency in Hz
|
|
* @dir: new clock direction - input/output.
|
|
*
|
|
* Configures the DAI master (MCLK) or system (SYSCLK) clocking.
|
|
*/
|
|
int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
|
|
unsigned int freq, int dir)
|
|
{
|
|
if (dai->ops && dai->ops->set_sysclk)
|
|
return dai->ops->set_sysclk(dai, clk_id, freq, dir);
|
|
else
|
|
return -EINVAL;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
|
|
|
|
/**
|
|
* snd_soc_dai_set_clkdiv - configure DAI clock dividers.
|
|
* @dai: DAI
|
|
* @div_id: DAI specific clock divider ID
|
|
* @div: new clock divisor.
|
|
*
|
|
* Configures the clock dividers. This is used to derive the best DAI bit and
|
|
* frame clocks from the system or master clock. It's best to set the DAI bit
|
|
* and frame clocks as low as possible to save system power.
|
|
*/
|
|
int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
|
|
int div_id, int div)
|
|
{
|
|
if (dai->ops && dai->ops->set_clkdiv)
|
|
return dai->ops->set_clkdiv(dai, div_id, div);
|
|
else
|
|
return -EINVAL;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv);
|
|
|
|
/**
|
|
* snd_soc_dai_set_pll - configure DAI PLL.
|
|
* @dai: DAI
|
|
* @pll_id: DAI specific PLL ID
|
|
* @source: DAI specific source for the PLL
|
|
* @freq_in: PLL input clock frequency in Hz
|
|
* @freq_out: requested PLL output clock frequency in Hz
|
|
*
|
|
* Configures and enables PLL to generate output clock based on input clock.
|
|
*/
|
|
int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source,
|
|
unsigned int freq_in, unsigned int freq_out)
|
|
{
|
|
if (dai->ops && dai->ops->set_pll)
|
|
return dai->ops->set_pll(dai, pll_id, source,
|
|
freq_in, freq_out);
|
|
else
|
|
return -EINVAL;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
|
|
|
|
/**
|
|
* snd_soc_dai_set_fmt - configure DAI hardware audio format.
|
|
* @dai: DAI
|
|
* @fmt: SND_SOC_DAIFMT_ format value.
|
|
*
|
|
* Configures the DAI hardware format and clocking.
|
|
*/
|
|
int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
|
|
{
|
|
if (dai->ops && dai->ops->set_fmt)
|
|
return dai->ops->set_fmt(dai, fmt);
|
|
else
|
|
return -EINVAL;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
|
|
|
|
/**
|
|
* snd_soc_dai_set_tdm_slot - configure DAI TDM.
|
|
* @dai: DAI
|
|
* @tx_mask: bitmask representing active TX slots.
|
|
* @rx_mask: bitmask representing active RX slots.
|
|
* @slots: Number of slots in use.
|
|
* @slot_width: Width in bits for each slot.
|
|
*
|
|
* Configures a DAI for TDM operation. Both mask and slots are codec and DAI
|
|
* specific.
|
|
*/
|
|
int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
|
|
unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width)
|
|
{
|
|
if (dai->ops && dai->ops->set_tdm_slot)
|
|
return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask,
|
|
slots, slot_width);
|
|
else
|
|
return -EINVAL;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
|
|
|
|
/**
|
|
* snd_soc_dai_set_channel_map - configure DAI audio channel map
|
|
* @dai: DAI
|
|
* @tx_num: how many TX channels
|
|
* @tx_slot: pointer to an array which imply the TX slot number channel
|
|
* 0~num-1 uses
|
|
* @rx_num: how many RX channels
|
|
* @rx_slot: pointer to an array which imply the RX slot number channel
|
|
* 0~num-1 uses
|
|
*
|
|
* configure the relationship between channel number and TDM slot number.
|
|
*/
|
|
int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
|
|
unsigned int tx_num, unsigned int *tx_slot,
|
|
unsigned int rx_num, unsigned int *rx_slot)
|
|
{
|
|
if (dai->ops && dai->ops->set_channel_map)
|
|
return dai->ops->set_channel_map(dai, tx_num, tx_slot,
|
|
rx_num, rx_slot);
|
|
else
|
|
return -EINVAL;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map);
|
|
|
|
/**
|
|
* snd_soc_dai_set_tristate - configure DAI system or master clock.
|
|
* @dai: DAI
|
|
* @tristate: tristate enable
|
|
*
|
|
* Tristates the DAI so that others can use it.
|
|
*/
|
|
int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate)
|
|
{
|
|
if (dai->ops && dai->ops->set_tristate)
|
|
return dai->ops->set_tristate(dai, tristate);
|
|
else
|
|
return -EINVAL;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
|
|
|
|
/**
|
|
* snd_soc_dai_digital_mute - configure DAI system or master clock.
|
|
* @dai: DAI
|
|
* @mute: mute enable
|
|
*
|
|
* Mutes the DAI DAC.
|
|
*/
|
|
int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute)
|
|
{
|
|
if (dai->ops && dai->ops->digital_mute)
|
|
return dai->ops->digital_mute(dai, mute);
|
|
else
|
|
return -EINVAL;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
|
|
|
|
/**
|
|
* snd_soc_register_card - Register a card with the ASoC core
|
|
*
|
|
* @card: Card to register
|
|
*
|
|
* Note that currently this is an internal only function: it will be
|
|
* exposed to machine drivers after further backporting of ASoC v2
|
|
* registration APIs.
|
|
*/
|
|
static int snd_soc_register_card(struct snd_soc_card *card)
|
|
{
|
|
if (!card->name || !card->dev)
|
|
return -EINVAL;
|
|
|
|
INIT_LIST_HEAD(&card->list);
|
|
card->instantiated = 0;
|
|
|
|
mutex_lock(&client_mutex);
|
|
list_add(&card->list, &card_list);
|
|
snd_soc_instantiate_cards();
|
|
mutex_unlock(&client_mutex);
|
|
|
|
dev_dbg(card->dev, "Registered card '%s'\n", card->name);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* snd_soc_unregister_card - Unregister a card with the ASoC core
|
|
*
|
|
* @card: Card to unregister
|
|
*
|
|
* Note that currently this is an internal only function: it will be
|
|
* exposed to machine drivers after further backporting of ASoC v2
|
|
* registration APIs.
|
|
*/
|
|
static int snd_soc_unregister_card(struct snd_soc_card *card)
|
|
{
|
|
mutex_lock(&client_mutex);
|
|
list_del(&card->list);
|
|
mutex_unlock(&client_mutex);
|
|
|
|
dev_dbg(card->dev, "Unregistered card '%s'\n", card->name);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* snd_soc_register_dai - Register a DAI with the ASoC core
|
|
*
|
|
* @dai: DAI to register
|
|
*/
|
|
int snd_soc_register_dai(struct snd_soc_dai *dai)
|
|
{
|
|
if (!dai->name)
|
|
return -EINVAL;
|
|
|
|
/* The device should become mandatory over time */
|
|
if (!dai->dev)
|
|
printk(KERN_WARNING "No device for DAI %s\n", dai->name);
|
|
|
|
if (!dai->ops)
|
|
dai->ops = &null_dai_ops;
|
|
|
|
INIT_LIST_HEAD(&dai->list);
|
|
|
|
mutex_lock(&client_mutex);
|
|
list_add(&dai->list, &dai_list);
|
|
snd_soc_instantiate_cards();
|
|
mutex_unlock(&client_mutex);
|
|
|
|
pr_debug("Registered DAI '%s'\n", dai->name);
|
|
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_register_dai);
|
|
|
|
/**
|
|
* snd_soc_unregister_dai - Unregister a DAI from the ASoC core
|
|
*
|
|
* @dai: DAI to unregister
|
|
*/
|
|
void snd_soc_unregister_dai(struct snd_soc_dai *dai)
|
|
{
|
|
mutex_lock(&client_mutex);
|
|
list_del(&dai->list);
|
|
mutex_unlock(&client_mutex);
|
|
|
|
pr_debug("Unregistered DAI '%s'\n", dai->name);
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_unregister_dai);
|
|
|
|
/**
|
|
* snd_soc_register_dais - Register multiple DAIs with the ASoC core
|
|
*
|
|
* @dai: Array of DAIs to register
|
|
* @count: Number of DAIs
|
|
*/
|
|
int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count)
|
|
{
|
|
int i, ret;
|
|
|
|
for (i = 0; i < count; i++) {
|
|
ret = snd_soc_register_dai(&dai[i]);
|
|
if (ret != 0)
|
|
goto err;
|
|
}
|
|
|
|
return 0;
|
|
|
|
err:
|
|
for (i--; i >= 0; i--)
|
|
snd_soc_unregister_dai(&dai[i]);
|
|
|
|
return ret;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_register_dais);
|
|
|
|
/**
|
|
* snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core
|
|
*
|
|
* @dai: Array of DAIs to unregister
|
|
* @count: Number of DAIs
|
|
*/
|
|
void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < count; i++)
|
|
snd_soc_unregister_dai(&dai[i]);
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_unregister_dais);
|
|
|
|
/**
|
|
* snd_soc_register_platform - Register a platform with the ASoC core
|
|
*
|
|
* @platform: platform to register
|
|
*/
|
|
int snd_soc_register_platform(struct snd_soc_platform *platform)
|
|
{
|
|
if (!platform->name)
|
|
return -EINVAL;
|
|
|
|
INIT_LIST_HEAD(&platform->list);
|
|
|
|
mutex_lock(&client_mutex);
|
|
list_add(&platform->list, &platform_list);
|
|
snd_soc_instantiate_cards();
|
|
mutex_unlock(&client_mutex);
|
|
|
|
pr_debug("Registered platform '%s'\n", platform->name);
|
|
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_register_platform);
|
|
|
|
/**
|
|
* snd_soc_unregister_platform - Unregister a platform from the ASoC core
|
|
*
|
|
* @platform: platform to unregister
|
|
*/
|
|
void snd_soc_unregister_platform(struct snd_soc_platform *platform)
|
|
{
|
|
mutex_lock(&client_mutex);
|
|
list_del(&platform->list);
|
|
mutex_unlock(&client_mutex);
|
|
|
|
pr_debug("Unregistered platform '%s'\n", platform->name);
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);
|
|
|
|
static u64 codec_format_map[] = {
|
|
SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE,
|
|
SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE,
|
|
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE,
|
|
SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE,
|
|
SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE,
|
|
SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE,
|
|
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3BE,
|
|
SNDRV_PCM_FMTBIT_U24_3LE | SNDRV_PCM_FMTBIT_U24_3BE,
|
|
SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE,
|
|
SNDRV_PCM_FMTBIT_U20_3LE | SNDRV_PCM_FMTBIT_U20_3BE,
|
|
SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE,
|
|
SNDRV_PCM_FMTBIT_U18_3LE | SNDRV_PCM_FMTBIT_U18_3BE,
|
|
SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE,
|
|
SNDRV_PCM_FMTBIT_FLOAT64_LE | SNDRV_PCM_FMTBIT_FLOAT64_BE,
|
|
SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE
|
|
| SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE,
|
|
};
|
|
|
|
/* Fix up the DAI formats for endianness: codecs don't actually see
|
|
* the endianness of the data but we're using the CPU format
|
|
* definitions which do need to include endianness so we ensure that
|
|
* codec DAIs always have both big and little endian variants set.
|
|
*/
|
|
static void fixup_codec_formats(struct snd_soc_pcm_stream *stream)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < ARRAY_SIZE(codec_format_map); i++)
|
|
if (stream->formats & codec_format_map[i])
|
|
stream->formats |= codec_format_map[i];
|
|
}
|
|
|
|
/**
|
|
* snd_soc_register_codec - Register a codec with the ASoC core
|
|
*
|
|
* @codec: codec to register
|
|
*/
|
|
int snd_soc_register_codec(struct snd_soc_codec *codec)
|
|
{
|
|
int i;
|
|
|
|
if (!codec->name)
|
|
return -EINVAL;
|
|
|
|
/* The device should become mandatory over time */
|
|
if (!codec->dev)
|
|
printk(KERN_WARNING "No device for codec %s\n", codec->name);
|
|
|
|
INIT_LIST_HEAD(&codec->list);
|
|
|
|
for (i = 0; i < codec->num_dai; i++) {
|
|
fixup_codec_formats(&codec->dai[i].playback);
|
|
fixup_codec_formats(&codec->dai[i].capture);
|
|
}
|
|
|
|
mutex_lock(&client_mutex);
|
|
list_add(&codec->list, &codec_list);
|
|
snd_soc_instantiate_cards();
|
|
mutex_unlock(&client_mutex);
|
|
|
|
pr_debug("Registered codec '%s'\n", codec->name);
|
|
|
|
return 0;
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_register_codec);
|
|
|
|
/**
|
|
* snd_soc_unregister_codec - Unregister a codec from the ASoC core
|
|
*
|
|
* @codec: codec to unregister
|
|
*/
|
|
void snd_soc_unregister_codec(struct snd_soc_codec *codec)
|
|
{
|
|
mutex_lock(&client_mutex);
|
|
list_del(&codec->list);
|
|
mutex_unlock(&client_mutex);
|
|
|
|
pr_debug("Unregistered codec '%s'\n", codec->name);
|
|
}
|
|
EXPORT_SYMBOL_GPL(snd_soc_unregister_codec);
|
|
|
|
static int __init snd_soc_init(void)
|
|
{
|
|
#ifdef CONFIG_DEBUG_FS
|
|
debugfs_root = debugfs_create_dir("asoc", NULL);
|
|
if (IS_ERR(debugfs_root) || !debugfs_root) {
|
|
printk(KERN_WARNING
|
|
"ASoC: Failed to create debugfs directory\n");
|
|
debugfs_root = NULL;
|
|
}
|
|
#endif
|
|
|
|
return platform_driver_register(&soc_driver);
|
|
}
|
|
|
|
static void __exit snd_soc_exit(void)
|
|
{
|
|
#ifdef CONFIG_DEBUG_FS
|
|
debugfs_remove_recursive(debugfs_root);
|
|
#endif
|
|
platform_driver_unregister(&soc_driver);
|
|
}
|
|
|
|
module_init(snd_soc_init);
|
|
module_exit(snd_soc_exit);
|
|
|
|
/* Module information */
|
|
MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
|
|
MODULE_DESCRIPTION("ALSA SoC Core");
|
|
MODULE_LICENSE("GPL");
|
|
MODULE_ALIAS("platform:soc-audio");
|