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-rw-r--r--recipes-kernel/aim-sound/files/sound.c768
1 files changed, 768 insertions, 0 deletions
diff --git a/recipes-kernel/aim-sound/files/sound.c b/recipes-kernel/aim-sound/files/sound.c
new file mode 100644
index 000000000..63b866b3c
--- /dev/null
+++ b/recipes-kernel/aim-sound/files/sound.c
@@ -0,0 +1,768 @@
+/*
+ * sound.c - Audio Application Interface Module for Mostcore
+ *
+ * Copyright (C) 2015 Microchip Technology Germany II GmbH & Co. KG
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * This file is licensed under GPLv2.
+ */
+
+#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
+
+#include <linux/module.h>
+#include <linux/printk.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <linux/sched.h>
+#include <linux/kthread.h>
+#include <mostcore.h>
+
+#define DRIVER_NAME "sound"
+
+static struct list_head dev_list;
+static struct most_aim audio_aim;
+
+/**
+ * struct channel - private structure to keep channel specific data
+ * @substream: stores the substream structure
+ * @iface: interface for which the channel belongs to
+ * @cfg: channel configuration
+ * @card: registered sound card
+ * @list: list for private use
+ * @id: channel index
+ * @period_pos: current period position (ring buffer)
+ * @buffer_pos: current buffer position (ring buffer)
+ * @is_stream_running: identifies whether a stream is running or not
+ * @opened: set when the stream is opened
+ * @playback_task: playback thread
+ * @playback_waitq: waitq used by playback thread
+ */
+struct channel {
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_hardware pcm_hardware;
+ struct most_interface *iface;
+ struct most_channel_config *cfg;
+ struct snd_card *card;
+ struct list_head list;
+ int id;
+ unsigned int period_pos;
+ unsigned int buffer_pos;
+ bool is_stream_running;
+
+ struct task_struct *playback_task;
+ wait_queue_head_t playback_waitq;
+
+ void (*copy_fn)(void *alsa, void *most, unsigned int bytes);
+};
+
+#define MOST_PCM_INFO (SNDRV_PCM_INFO_MMAP | \
+ SNDRV_PCM_INFO_MMAP_VALID | \
+ SNDRV_PCM_INFO_BATCH | \
+ SNDRV_PCM_INFO_INTERLEAVED | \
+ SNDRV_PCM_INFO_BLOCK_TRANSFER)
+
+#define swap16(val) ( \
+ (((u16)(val) << 8) & (u16)0xFF00) | \
+ (((u16)(val) >> 8) & (u16)0x00FF))
+
+#define swap32(val) ( \
+ (((u32)(val) << 24) & (u32)0xFF000000) | \
+ (((u32)(val) << 8) & (u32)0x00FF0000) | \
+ (((u32)(val) >> 8) & (u32)0x0000FF00) | \
+ (((u32)(val) >> 24) & (u32)0x000000FF))
+
+static void swap_copy16(u16 *dest, const u16 *source, unsigned int bytes)
+{
+ unsigned int i = 0;
+
+ while (i < (bytes / 2)) {
+ dest[i] = swap16(source[i]);
+ i++;
+ }
+}
+
+static void swap_copy24(u8 *dest, const u8 *source, unsigned int bytes)
+{
+ unsigned int i = 0;
+
+ while (i < bytes - 2) {
+ dest[i] = source[i + 2];
+ dest[i + 1] = source[i + 1];
+ dest[i + 2] = source[i];
+ i += 3;
+ }
+}
+
+static void swap_copy32(u32 *dest, const u32 *source, unsigned int bytes)
+{
+ unsigned int i = 0;
+
+ while (i < bytes / 4) {
+ dest[i] = swap32(source[i]);
+ i++;
+ }
+}
+
+static void alsa_to_most_memcpy(void *alsa, void *most, unsigned int bytes)
+{
+ memcpy(most, alsa, bytes);
+}
+
+static void alsa_to_most_copy16(void *alsa, void *most, unsigned int bytes)
+{
+ swap_copy16(most, alsa, bytes);
+}
+
+static void alsa_to_most_copy24(void *alsa, void *most, unsigned int bytes)
+{
+ swap_copy24(most, alsa, bytes);
+}
+
+static void alsa_to_most_copy32(void *alsa, void *most, unsigned int bytes)
+{
+ swap_copy32(most, alsa, bytes);
+}
+
+static void most_to_alsa_memcpy(void *alsa, void *most, unsigned int bytes)
+{
+ memcpy(alsa, most, bytes);
+}
+
+static void most_to_alsa_copy16(void *alsa, void *most, unsigned int bytes)
+{
+ swap_copy16(alsa, most, bytes);
+}
+
+static void most_to_alsa_copy24(void *alsa, void *most, unsigned int bytes)
+{
+ swap_copy24(alsa, most, bytes);
+}
+
+static void most_to_alsa_copy32(void *alsa, void *most, unsigned int bytes)
+{
+ swap_copy32(alsa, most, bytes);
+}
+
+/**
+ * get_channel - get pointer to channel
+ * @iface: interface structure
+ * @channel_id: channel ID
+ *
+ * This traverses the channel list and returns the channel matching the
+ * ID and interface.
+ *
+ * Returns pointer to channel on success or NULL otherwise.
+ */
+static struct channel *get_channel(struct most_interface *iface,
+ int channel_id)
+{
+ struct channel *channel, *tmp;
+
+ list_for_each_entry_safe(channel, tmp, &dev_list, list) {
+ if ((channel->iface == iface) && (channel->id == channel_id))
+ return channel;
+ }
+
+ return NULL;
+}
+
+/**
+ * copy_data - implements data copying function
+ * @channel: channel
+ * @mbo: MBO from core
+ *
+ * Copy data from/to ring buffer to/from MBO and update the buffer position
+ */
+static bool copy_data(struct channel *channel, struct mbo *mbo)
+{
+ struct snd_pcm_runtime *const runtime = channel->substream->runtime;
+ unsigned int const frame_bytes = channel->cfg->subbuffer_size;
+ unsigned int const buffer_size = runtime->buffer_size;
+ unsigned int frames;
+ unsigned int fr0;
+
+ if (channel->cfg->direction & MOST_CH_RX)
+ frames = mbo->processed_length / frame_bytes;
+ else
+ frames = mbo->buffer_length / frame_bytes;
+ fr0 = min(buffer_size - channel->buffer_pos, frames);
+
+ channel->copy_fn(runtime->dma_area + channel->buffer_pos * frame_bytes,
+ mbo->virt_address,
+ fr0 * frame_bytes);
+
+ if (frames > fr0) {
+ /* wrap around at end of ring buffer */
+ channel->copy_fn(runtime->dma_area,
+ mbo->virt_address + fr0 * frame_bytes,
+ (frames - fr0) * frame_bytes);
+ }
+
+ channel->buffer_pos += frames;
+ if (channel->buffer_pos >= buffer_size)
+ channel->buffer_pos -= buffer_size;
+ channel->period_pos += frames;
+ if (channel->period_pos >= runtime->period_size) {
+ channel->period_pos -= runtime->period_size;
+ return true;
+ }
+
+ return false;
+}
+
+/**
+ * playback_thread - function implements the playback thread
+ * @data: private data
+ *
+ * Thread which does the playback functionality in a loop. It waits for a free
+ * MBO from mostcore for a particular channel and copy the data from ring buffer
+ * to MBO. Submit the MBO back to mostcore, after copying the data.
+ *
+ * Returns 0 on success or error code otherwise.
+ */
+static int playback_thread(void *data)
+{
+ struct channel *const channel = data;
+
+ while (!kthread_should_stop()) {
+ struct mbo *mbo = NULL;
+ bool period_elapsed = false;
+ int ret;
+
+ wait_event_interruptible(
+ channel->playback_waitq,
+ kthread_should_stop() ||
+ (channel->is_stream_running &&
+ (mbo = most_get_mbo(channel->iface, channel->id,
+ &audio_aim))));
+ if (!mbo)
+ continue;
+
+ if (channel->is_stream_running)
+ period_elapsed = copy_data(channel, mbo);
+ else
+ memset(mbo->virt_address, 0, mbo->buffer_length);
+
+ ret = most_submit_mbo(mbo);
+ if (ret)
+ channel->is_stream_running = false;
+
+ if (period_elapsed)
+ snd_pcm_period_elapsed(channel->substream);
+ }
+
+ return 0;
+}
+
+/**
+ * pcm_open - implements open callback function for PCM middle layer
+ * @substream: pointer to ALSA PCM substream
+ *
+ * This is called when a PCM substream is opened. At least, the function should
+ * initialize the runtime->hw record.
+ *
+ * Returns 0 on success or error code otherwise.
+ */
+static int pcm_open(struct snd_pcm_substream *substream)
+{
+ struct channel *channel = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct most_channel_config *cfg = channel->cfg;
+
+ channel->substream = substream;
+
+ if (cfg->direction == MOST_CH_TX) {
+ channel->playback_task = kthread_run(playback_thread, channel,
+ "most_audio_playback");
+ if (IS_ERR(channel->playback_task)) {
+ pr_err("Couldn't start thread\n");
+ return PTR_ERR(channel->playback_task);
+ }
+ }
+
+ if (most_start_channel(channel->iface, channel->id, &audio_aim)) {
+ pr_err("most_start_channel() failed!\n");
+ if (cfg->direction == MOST_CH_TX)
+ kthread_stop(channel->playback_task);
+ return -EBUSY;
+ }
+
+ runtime->hw = channel->pcm_hardware;
+ return 0;
+}
+
+/**
+ * pcm_close - implements close callback function for PCM middle layer
+ * @substream: sub-stream pointer
+ *
+ * Obviously, this is called when a PCM substream is closed. Any private
+ * instance for a PCM substream allocated in the open callback will be
+ * released here.
+ *
+ * Returns 0 on success or error code otherwise.
+ */
+static int pcm_close(struct snd_pcm_substream *substream)
+{
+ struct channel *channel = substream->private_data;
+
+ if (channel->cfg->direction == MOST_CH_TX)
+ kthread_stop(channel->playback_task);
+ most_stop_channel(channel->iface, channel->id, &audio_aim);
+
+ return 0;
+}
+
+/**
+ * pcm_hw_params - implements hw_params callback function for PCM middle layer
+ * @substream: sub-stream pointer
+ * @hw_params: contains the hardware parameters set by the application
+ *
+ * This is called when the hardware parameters is set by the application, that
+ * is, once when the buffer size, the period size, the format, etc. are defined
+ * for the PCM substream. Many hardware setups should be done is this callback,
+ * including the allocation of buffers.
+ *
+ * Returns 0 on success or error code otherwise.
+ */
+static int pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct channel *channel = substream->private_data;
+
+ if ((params_channels(hw_params) > channel->pcm_hardware.channels_max) ||
+ (params_channels(hw_params) < channel->pcm_hardware.channels_min)) {
+ pr_err("Requested number of channels not supported.\n");
+ return -EINVAL;
+ }
+ return snd_pcm_lib_alloc_vmalloc_buffer(substream,
+ params_buffer_bytes(hw_params));
+}
+
+/**
+ * pcm_hw_free - implements hw_free callback function for PCM middle layer
+ * @substream: substream pointer
+ *
+ * This is called to release the resources allocated via hw_params.
+ * This function will be always called before the close callback is called.
+ *
+ * Returns 0 on success or error code otherwise.
+ */
+static int pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+/**
+ * pcm_prepare - implements prepare callback function for PCM middle layer
+ * @substream: substream pointer
+ *
+ * This callback is called when the PCM is "prepared". Format rate, sample rate,
+ * etc., can be set here. This callback can be called many times at each setup.
+ *
+ * Returns 0 on success or error code otherwise.
+ */
+static int pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct channel *channel = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct most_channel_config *cfg = channel->cfg;
+ int width = snd_pcm_format_physical_width(runtime->format);
+
+ channel->copy_fn = NULL;
+
+ if (cfg->direction == MOST_CH_TX) {
+ if (snd_pcm_format_big_endian(runtime->format) || width == 8)
+ channel->copy_fn = alsa_to_most_memcpy;
+ else if (width == 16)
+ channel->copy_fn = alsa_to_most_copy16;
+ else if (width == 24)
+ channel->copy_fn = alsa_to_most_copy24;
+ else if (width == 32)
+ channel->copy_fn = alsa_to_most_copy32;
+ } else {
+ if (snd_pcm_format_big_endian(runtime->format) || width == 8)
+ channel->copy_fn = most_to_alsa_memcpy;
+ else if (width == 16)
+ channel->copy_fn = most_to_alsa_copy16;
+ else if (width == 24)
+ channel->copy_fn = most_to_alsa_copy24;
+ else if (width == 32)
+ channel->copy_fn = most_to_alsa_copy32;
+ }
+
+ if (!channel->copy_fn) {
+ pr_err("unsupported format\n");
+ return -EINVAL;
+ }
+
+ channel->period_pos = 0;
+ channel->buffer_pos = 0;
+
+ return 0;
+}
+
+/**
+ * pcm_trigger - implements trigger callback function for PCM middle layer
+ * @substream: substream pointer
+ * @cmd: action to perform
+ *
+ * This is called when the PCM is started, stopped or paused. The action will be
+ * specified in the second argument, SNDRV_PCM_TRIGGER_XXX
+ *
+ * Returns 0 on success or error code otherwise.
+ */
+static int pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct channel *channel = substream->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ channel->is_stream_running = true;
+ wake_up_interruptible(&channel->playback_waitq);
+ return 0;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ channel->is_stream_running = false;
+ return 0;
+
+ default:
+ pr_info("pcm_trigger(), invalid\n");
+ return -EINVAL;
+ }
+ return 0;
+}
+
+/**
+ * pcm_pointer - implements pointer callback function for PCM middle layer
+ * @substream: substream pointer
+ *
+ * This callback is called when the PCM middle layer inquires the current
+ * hardware position on the buffer. The position must be returned in frames,
+ * ranging from 0 to buffer_size-1.
+ */
+static snd_pcm_uframes_t pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct channel *channel = substream->private_data;
+
+ return channel->buffer_pos;
+}
+
+/**
+ * Initialization of struct snd_pcm_ops
+ */
+static struct snd_pcm_ops pcm_ops = {
+ .open = pcm_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_hw_params,
+ .hw_free = pcm_hw_free,
+ .prepare = pcm_prepare,
+ .trigger = pcm_trigger,
+ .pointer = pcm_pointer,
+ .page = snd_pcm_lib_get_vmalloc_page,
+ .mmap = snd_pcm_lib_mmap_vmalloc,
+};
+
+static int split_arg_list(char *buf, char **card_name, char **pcm_format)
+{
+ *card_name = strsep(&buf, ".");
+ if (!*card_name)
+ return -EIO;
+ *pcm_format = strsep(&buf, ".\n");
+ if (!*pcm_format)
+ return -EIO;
+ return 0;
+}
+
+static int audio_set_hw_params(struct snd_pcm_hardware *pcm_hw,
+ char *pcm_format,
+ struct most_channel_config *cfg)
+{
+ pcm_hw->info = MOST_PCM_INFO;
+ pcm_hw->rates = SNDRV_PCM_RATE_48000;
+ pcm_hw->rate_min = 48000;
+ pcm_hw->rate_max = 48000;
+ pcm_hw->buffer_bytes_max = cfg->num_buffers * cfg->buffer_size;
+ pcm_hw->period_bytes_min = cfg->buffer_size;
+ pcm_hw->period_bytes_max = cfg->buffer_size;
+ pcm_hw->periods_min = 1;
+ pcm_hw->periods_max = cfg->num_buffers;
+
+ if (!strcmp(pcm_format, "1x8")) {
+ if (cfg->subbuffer_size != 1)
+ goto error;
+ pr_info("PCM format is 8-bit mono\n");
+ pcm_hw->channels_min = 1;
+ pcm_hw->channels_max = 1;
+ pcm_hw->formats = SNDRV_PCM_FMTBIT_S8;
+ } else if (!strcmp(pcm_format, "2x16")) {
+ if (cfg->subbuffer_size != 4)
+ goto error;
+ pr_info("PCM format is 16-bit stereo\n");
+ pcm_hw->channels_min = 2;
+ pcm_hw->channels_max = 2;
+ pcm_hw->formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S16_BE;
+ } else if (!strcmp(pcm_format, "2x24")) {
+ if (cfg->subbuffer_size != 6)
+ goto error;
+ pr_info("PCM format is 24-bit stereo\n");
+ pcm_hw->channels_min = 2;
+ pcm_hw->channels_max = 2;
+ pcm_hw->formats = SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S24_3BE;
+ } else if (!strcmp(pcm_format, "2x32")) {
+ if (cfg->subbuffer_size != 8)
+ goto error;
+ pr_info("PCM format is 32-bit stereo\n");
+ pcm_hw->channels_min = 2;
+ pcm_hw->channels_max = 2;
+ pcm_hw->formats = SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_S32_BE;
+ } else if (!strcmp(pcm_format, "6x16")) {
+ if (cfg->subbuffer_size != 12)
+ goto error;
+ pr_info("PCM format is 16-bit 5.1 multi channel\n");
+ pcm_hw->channels_min = 6;
+ pcm_hw->channels_max = 6;
+ pcm_hw->formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S16_BE;
+ } else {
+ pr_err("PCM format %s not supported\n", pcm_format);
+ return -EIO;
+ }
+ return 0;
+error:
+ pr_err("Audio resolution doesn't fit subbuffer size\n");
+ return -EINVAL;
+}
+
+/**
+ * audio_probe_channel - probe function of the driver module
+ * @iface: pointer to interface instance
+ * @channel_id: channel index/ID
+ * @cfg: pointer to actual channel configuration
+ * @parent: pointer to kobject (needed for sysfs hook-up)
+ * @arg_list: string that provides the name of the device to be created in /dev
+ * plus the desired audio resolution
+ *
+ * Creates sound card, pcm device, sets pcm ops and registers sound card.
+ *
+ * Returns 0 on success or error code otherwise.
+ */
+static int audio_probe_channel(struct most_interface *iface, int channel_id,
+ struct most_channel_config *cfg,
+ struct kobject *parent, char *arg_list)
+{
+ struct channel *channel;
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ int playback_count = 0;
+ int capture_count = 0;
+ int ret;
+ int direction;
+ char *card_name;
+ char *pcm_format;
+
+ if (!iface)
+ return -EINVAL;
+
+ if (cfg->data_type != MOST_CH_SYNC) {
+ pr_err("Incompatible channel type\n");
+ return -EINVAL;
+ }
+
+ if (get_channel(iface, channel_id)) {
+ pr_err("channel (%s:%d) is already linked\n",
+ iface->description, channel_id);
+ return -EINVAL;
+ }
+
+ if (cfg->direction == MOST_CH_TX) {
+ playback_count = 1;
+ direction = SNDRV_PCM_STREAM_PLAYBACK;
+ } else {
+ capture_count = 1;
+ direction = SNDRV_PCM_STREAM_CAPTURE;
+ }
+
+ ret = split_arg_list(arg_list, &card_name, &pcm_format);
+ if (ret < 0) {
+ pr_info("PCM format missing\n");
+ return ret;
+ }
+
+ ret = snd_card_create(-1, card_name, THIS_MODULE,
+- sizeof(*channel), &card);
+ if (ret < 0)
+ return ret;
+
+ channel = card->private_data;
+ channel->card = card;
+ channel->cfg = cfg;
+ channel->iface = iface;
+ channel->id = channel_id;
+ init_waitqueue_head(&channel->playback_waitq);
+
+ if (audio_set_hw_params(&channel->pcm_hardware, pcm_format, cfg))
+ goto err_free_card;
+
+ snprintf(card->driver, sizeof(card->driver), "%s", DRIVER_NAME);
+ snprintf(card->shortname, sizeof(card->shortname), "Microchip MOST:%d",
+ card->number);
+ snprintf(card->longname, sizeof(card->longname), "%s at %s, ch %d",
+ card->shortname, iface->description, channel_id);
+
+ ret = snd_pcm_new(card, card_name, 0, playback_count,
+ capture_count, &pcm);
+ if (ret < 0)
+ goto err_free_card;
+
+ pcm->private_data = channel;
+
+ snd_pcm_set_ops(pcm, direction, &pcm_ops);
+
+ ret = snd_card_register(card);
+ if (ret < 0)
+ goto err_free_card;
+
+ list_add_tail(&channel->list, &dev_list);
+
+ return 0;
+
+err_free_card:
+ snd_card_free(card);
+ return ret;
+}
+
+/**
+ * audio_disconnect_channel - function to disconnect a channel
+ * @iface: pointer to interface instance
+ * @channel_id: channel index
+ *
+ * This frees allocated memory and removes the sound card from ALSA
+ *
+ * Returns 0 on success or error code otherwise.
+ */
+static int audio_disconnect_channel(struct most_interface *iface,
+ int channel_id)
+{
+ struct channel *channel;
+
+ channel = get_channel(iface, channel_id);
+ if (!channel) {
+ pr_err("sound_disconnect_channel(), invalid channel %d\n",
+ channel_id);
+ return -EINVAL;
+ }
+
+ list_del(&channel->list);
+ snd_card_free(channel->card);
+
+ return 0;
+}
+
+/**
+ * audio_rx_completion - completion handler for rx channels
+ * @mbo: pointer to buffer object that has completed
+ *
+ * This searches for the channel this MBO belongs to and copy the data from MBO
+ * to ring buffer
+ *
+ * Returns 0 on success or error code otherwise.
+ */
+static int audio_rx_completion(struct mbo *mbo)
+{
+ struct channel *channel = get_channel(mbo->ifp, mbo->hdm_channel_id);
+ bool period_elapsed = false;
+
+ if (!channel) {
+ pr_err("sound_rx_completion(), invalid channel %d\n",
+ mbo->hdm_channel_id);
+ return -EINVAL;
+ }
+
+ if (channel->is_stream_running)
+ period_elapsed = copy_data(channel, mbo);
+
+ most_put_mbo(mbo);
+
+ if (period_elapsed)
+ snd_pcm_period_elapsed(channel->substream);
+
+ return 0;
+}
+
+/**
+ * audio_tx_completion - completion handler for tx channels
+ * @iface: pointer to interface instance
+ * @channel_id: channel index/ID
+ *
+ * This searches the channel that belongs to this combination of interface
+ * pointer and channel ID and wakes a process sitting in the wait queue of
+ * this channel.
+ *
+ * Returns 0 on success or error code otherwise.
+ */
+static int audio_tx_completion(struct most_interface *iface, int channel_id)
+{
+ struct channel *channel = get_channel(iface, channel_id);
+
+ if (!channel) {
+ pr_err("sound_tx_completion(), invalid channel %d\n",
+ channel_id);
+ return -EINVAL;
+ }
+
+ wake_up_interruptible(&channel->playback_waitq);
+
+ return 0;
+}
+
+/**
+ * Initialization of the struct most_aim
+ */
+static struct most_aim audio_aim = {
+ .name = DRIVER_NAME,
+ .probe_channel = audio_probe_channel,
+ .disconnect_channel = audio_disconnect_channel,
+ .rx_completion = audio_rx_completion,
+ .tx_completion = audio_tx_completion,
+};
+
+static int __init audio_init(void)
+{
+ pr_info("init()\n");
+
+ INIT_LIST_HEAD(&dev_list);
+
+ return most_register_aim(&audio_aim);
+}
+
+static void __exit audio_exit(void)
+{
+ struct channel *channel, *tmp;
+
+ pr_info("exit()\n");
+
+ list_for_each_entry_safe(channel, tmp, &dev_list, list) {
+ list_del(&channel->list);
+ snd_card_free(channel->card);
+ }
+
+ most_deregister_aim(&audio_aim);
+}
+
+module_init(audio_init);
+module_exit(audio_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_AUTHOR("Christian Gromm <christian.gromm@microchip.com>");
+MODULE_DESCRIPTION("Audio Application Interface Module for MostCore");