/* * soc-core.c -- ALSA SoC Audio Layer * * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * * Author: Liam Girdwood * with code, comments and ideas from :- * Richard Purdie * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * * TODO: * o Add hw rules to enforce rates, etc. * o More testing with other codecs/machines. * o Add more codecs and platforms to ensure good API coverage. * o Support TDM on PCM and I2S */ #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include static DEFINE_MUTEX(pcm_mutex); static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); #ifdef CONFIG_DEBUG_FS static struct dentry *debugfs_root; #endif static DEFINE_MUTEX(client_mutex); static LIST_HEAD(card_list); static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); static int snd_soc_register_card(struct snd_soc_card *card); static int snd_soc_unregister_card(struct snd_soc_card *card); /* * This is a timeout to do a DAPM powerdown after a stream is closed(). * It can be used to eliminate pops between different playback streams, e.g. * between two audio tracks. */ static int pmdown_time = 5000; module_param(pmdown_time, int, 0); MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); /* * This function forces any delayed work to be queued and run. */ static int run_delayed_work(struct delayed_work *dwork) { int ret; /* cancel any work waiting to be queued. */ ret = cancel_delayed_work(dwork); /* if there was any work waiting then we run it now and * wait for it's completion */ if (ret) { schedule_delayed_work(dwork, 0); flush_scheduled_work(); } return ret; } /* codec register dump */ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) { int i, step = 1, count = 0; if (!codec->reg_cache_size) return 0; if (codec->reg_cache_step) step = codec->reg_cache_step; count += sprintf(buf, "%s registers\n", codec->name); for (i = 0; i < codec->reg_cache_size; i += step) { if (codec->readable_register && !codec->readable_register(i)) continue; count += sprintf(buf + count, "%2x: ", i); if (count >= PAGE_SIZE - 1) break; if (codec->display_register) count += codec->display_register(codec, buf + count, PAGE_SIZE - count, i); else count += snprintf(buf + count, PAGE_SIZE - count, "%4x", codec->read(codec, i)); if (count >= PAGE_SIZE - 1) break; count += snprintf(buf + count, PAGE_SIZE - count, "\n"); if (count >= PAGE_SIZE - 1) break; } /* Truncate count; min() would cause a warning */ if (count >= PAGE_SIZE) count = PAGE_SIZE - 1; return count; } static ssize_t codec_reg_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_soc_device *devdata = dev_get_drvdata(dev); return soc_codec_reg_show(devdata->card->codec, buf); } static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); static ssize_t pmdown_time_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_soc_device *socdev = dev_get_drvdata(dev); struct snd_soc_card *card = socdev->card; return sprintf(buf, "%ld\n", card->pmdown_time); } static ssize_t pmdown_time_set(struct device *dev, struct device_attribute *attr, const char *buf, size_t count) { struct snd_soc_device *socdev = dev_get_drvdata(dev); struct snd_soc_card *card = socdev->card; strict_strtol(buf, 10, &card->pmdown_time); return count; } static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set); #ifdef CONFIG_DEBUG_FS static int codec_reg_open_file(struct inode *inode, struct file *file) { file->private_data = inode->i_private; return 0; } static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) { ssize_t ret; struct snd_soc_codec *codec = file->private_data; char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); if (!buf) return -ENOMEM; ret = soc_codec_reg_show(codec, buf); if (ret >= 0) ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); kfree(buf); return ret; } static ssize_t codec_reg_write_file(struct file *file, const char __user *user_buf, size_t count, loff_t *ppos) { char buf[32]; int buf_size; char *start = buf; unsigned long reg, value; int step = 1; struct snd_soc_codec *codec = file->private_data; buf_size = min(count, (sizeof(buf)-1)); if (copy_from_user(buf, user_buf, buf_size)) return -EFAULT; buf[buf_size] = 0; if (codec->reg_cache_step) step = codec->reg_cache_step; while (*start == ' ') start++; reg = simple_strtoul(start, &start, 16); if ((reg >= codec->reg_cache_size) || (reg % step)) return -EINVAL; while (*start == ' ') start++; if (strict_strtoul(start, 16, &value)) return -EINVAL; codec->write(codec, reg, value); return buf_size; } static const struct file_operations codec_reg_fops = { .open = codec_reg_open_file, .read = codec_reg_read_file, .write = codec_reg_write_file, }; static void soc_init_codec_debugfs(struct snd_soc_codec *codec) { char codec_root[128]; if (codec->dev) snprintf(codec_root, sizeof(codec_root), "%s.%s", codec->name, dev_name(codec->dev)); else snprintf(codec_root, sizeof(codec_root), "%s", codec->name); codec->debugfs_codec_root = debugfs_create_dir(codec_root, debugfs_root); if (!codec->debugfs_codec_root) { printk(KERN_WARNING "ASoC: Failed to create codec debugfs directory\n"); return; } codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, codec->debugfs_codec_root, codec, &codec_reg_fops); if (!codec->debugfs_reg) printk(KERN_WARNING "ASoC: Failed to create codec register debugfs file\n"); codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744, codec->debugfs_codec_root, &codec->pop_time); if (!codec->debugfs_pop_time) printk(KERN_WARNING "Failed to create pop time debugfs file\n"); codec->debugfs_dapm = debugfs_create_dir("dapm", codec->debugfs_codec_root); if (!codec->debugfs_dapm) printk(KERN_WARNING "Failed to create DAPM debugfs directory\n"); snd_soc_dapm_debugfs_init(codec); } static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) { debugfs_remove_recursive(codec->debugfs_codec_root); } #else static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec) { } static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) { } #endif #ifdef CONFIG_SND_SOC_AC97_BUS /* unregister ac97 codec */ static int soc_ac97_dev_unregister(struct snd_soc_codec *codec) { if (codec->ac97->dev.bus) device_unregister(&codec->ac97->dev); return 0; } /* stop no dev release warning */ static void soc_ac97_device_release(struct device *dev){} /* register ac97 codec to bus */ static int soc_ac97_dev_register(struct snd_soc_codec *codec) { int err; codec->ac97->dev.bus = &ac97_bus_type; codec->ac97->dev.parent = codec->card->dev; codec->ac97->dev.release = soc_ac97_device_release; dev_set_name(&codec->ac97->dev, "%d-%d:%s", codec->card->number, 0, codec->name); err = device_register(&codec->ac97->dev); if (err < 0) { snd_printk(KERN_ERR "Can't register ac97 bus\n"); codec->ac97->dev.bus = NULL; return err; } return 0; } #endif static int soc_pcm_apply_symmetry(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_dai *cpu_dai = machine->cpu_dai; struct snd_soc_dai *codec_dai = machine->codec_dai; int ret; if (codec_dai->symmetric_rates || cpu_dai->symmetric_rates || machine->symmetric_rates) { dev_dbg(card->dev, "Symmetry forces %dHz rate\n", machine->rate); ret = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, machine->rate, machine->rate); if (ret < 0) { dev_err(card->dev, "Unable to apply rate symmetry constraint: %d\n", ret); return ret; } } return 0; } /* * Called by ALSA when a PCM substream is opened, the runtime->hw record is * then initialized and any private data can be allocated. This also calls * startup for the cpu DAI, platform, machine and codec DAI. */ static int soc_pcm_open(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_pcm_runtime *runtime = substream->runtime; 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 = 0; mutex_lock(&pcm_mutex); /* startup the audio subsystem */ if (cpu_dai->ops->startup) { ret = cpu_dai->ops->startup(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open interface %s\n", cpu_dai->name); goto out; } } if (platform->pcm_ops->open) { ret = platform->pcm_ops->open(substream); if (ret < 0) { printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); goto platform_err; } } if (codec_dai->ops->startup) { ret = codec_dai->ops->startup(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open codec %s\n", codec_dai->name); goto codec_dai_err; } } if (machine->ops && machine->ops->startup) { ret = machine->ops->startup(substream); if (ret < 0) { printk(KERN_ERR "asoc: %s startup failed\n", machine->name); goto machine_err; } } /* Check that the codec and cpu DAI's are compatible */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { runtime->hw.rate_min = max(codec_dai->playback.rate_min, cpu_dai->playback.rate_min); runtime->hw.rate_max = min(codec_dai->playback.rate_max, cpu_dai->playback.rate_max); runtime->hw.channels_min = max(codec_dai->playback.channels_min, cpu_dai->playback.channels_min); runtime->hw.channels_max = min(codec_dai->playback.channels_max, cpu_dai->playback.channels_max); runtime->hw.formats = codec_dai->playback.formats & cpu_dai->playback.formats; runtime->hw.rates = codec_dai->playback.rates & cpu_dai->playback.rates; } else { runtime->hw.rate_min = max(codec_dai->capture.rate_min, cpu_dai->capture.rate_min); runtime->hw.rate_max = min(codec_dai->capture.rate_max, cpu_dai->capture.rate_max); runtime->hw.channels_min = 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; } 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 machine_err; } if (!runtime->hw.formats) { printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", codec_dai->name, cpu_dai->name); goto machine_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 machine_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 machine_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; machine_err: if (machine->ops && machine->ops->shutdown) machine->ops->shutdown(substream); 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 (dai->ops->digital_mute && dai->playback.active) dai->ops->digital_mute(dai, 1); } /* suspend all pcms */ for (i = 0; i < card->num_links; i++) 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 (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 (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 (codec_dev->suspend) codec_dev->suspend(pdev, PMSG_SUSPEND); for (i = 0; i < card->num_links; i++) { struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; 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"); 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 (cpu_dai->resume && cpu_dai->ac97_control) cpu_dai->resume(cpu_dai); } if (codec_dev->resume) codec_dev->resume(pdev); for (i = 0; i < codec->num_dai; i++) { char *stream = codec->dai[i].playback.stream_name; 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 (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 (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; /* 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 uninitalised 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) return 0; 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; } if (card->dai_link[i].codec_dai->ac97_control) { 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 max = mc->max; unsigned int shift = mc->shift; unsigned int rshift = mc->rshift; 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 = shift == rshift ? 1 : 2; uinfo->value.integer.min = 0; uinfo->value.integer.max = 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 max = mc->max; 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 = 2; uinfo->value.integer.min = 0; uinfo->value.integer.max = 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 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_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_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");